diff options
| -rw-r--r-- | ChangeLog | 36 | ||||
| -rw-r--r-- | docs/plugins/Makefile.am | 1 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins-docs.sgml | 1 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins-sections.txt | 10 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins.args | 20 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-audiofx.xml | 7 | ||||
| -rw-r--r-- | gst/audiofx/Makefile.am | 6 | ||||
| -rw-r--r-- | gst/audiofx/audioamplify.c | 451 | ||||
| -rw-r--r-- | gst/audiofx/audioamplify.h | 61 | ||||
| -rw-r--r-- | gst/audiofx/audiofx.c | 5 | ||||
| -rw-r--r-- | tests/check/Makefile.am | 1 | ||||
| -rw-r--r-- | tests/check/elements/audioamplify.c | 469 |
12 files changed, 1065 insertions, 3 deletions
@@ -1,3 +1,39 @@ +2007-01-24 Sebastian Dröge <slomo@circular-chaos.org> + + reviewed by: Stefan Kost <ensonic@users.sf.net> + + * gst/audiofx/Makefile.am: + * gst/audiofx/audioamplify.c: + (gst_audio_amplify_clipping_method_get_type), + (gst_audio_amplify_base_init), (gst_audio_amplify_class_init), + (gst_audio_amplify_init), (gst_audio_amplify_set_process_function), + (gst_audio_amplify_set_property), (gst_audio_amplify_get_property), + (gst_audio_amplify_set_caps), + (gst_audio_amplify_transform_int_clip), + (gst_audio_amplify_transform_int_wrap_negative), + (gst_audio_amplify_transform_int_wrap_positive), + (gst_audio_amplify_transform_float_clip), + (gst_audio_amplify_transform_float_wrap_negative), + (gst_audio_amplify_transform_float_wrap_positive), + (gst_audio_amplify_transform_ip): + * gst/audiofx/audioamplify.h: + * gst/audiofx/audiofx.c: (plugin_init): + Add new element "audioamplify". This allows scaling of raw audio + samples, similar to the "volume" element, but provides different modes + for clipping and allows unlimited amplification. It's mainly targeted + for creative sound design and not as a replacement of the "volume" + element. Fixes #397162 + * 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-audiofx.xml: + Add docs for audioamplify and integrate them into the build system + * tests/check/Makefile.am: + * tests/check/elements/audioamplify.c: (setup_amplify), + (cleanup_amplify), (GST_START_TEST), (amplify_suite), (main): + Add fairly extensive unit test suite for audioamplify + 2007-01-24 Wim Taymans <wim@fluendo.com> * gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked): diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index e7746e30..2a758478 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -77,6 +77,7 @@ EXTRA_HFILES = \ $(top_srcdir)/gst/apetag/gstapedemux.h \ $(top_srcdir)/gst/audiofx/audiopanorama.h \ $(top_srcdir)/gst/audiofx/audioinvert.h \ + $(top_srcdir)/gst/audiofx/audioamplify.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 5f7af731..39b97578 100644 --- a/docs/plugins/gst-plugins-good-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml @@ -16,6 +16,7 @@ <xi:include href="xml/element-apev2mux.xml" /> <xi:include href="xml/element-audiopanorama.xml" /> <xi:include href="xml/element-audioinvert.xml" /> + <xi:include href="xml/element-audioamplify.xml" /> <xi:include href="xml/element-autoaudiosink.xml" /> <xi:include href="xml/element-autovideosink.xml" /> <xi:include href="xml/element-avidemux.xml" /> diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index 9da8dacf..bb8d1331 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -48,6 +48,16 @@ GST_AUDIO_INVERT_CLASS </SECTION> <SECTION> +<FILE>element-audioamplify</FILE> +GstAudioAmplify +<TITLE>audioamplify</TITLE> +<SUBSECTION Standard> +GstAudioAmplifyClass +GST_AUDIO_AMPLIFY +GST_AUDIO_AMPLIFY_CLASS +</SECTION> + +<SECTION> <FILE>element-autoaudiosink</FILE> GstAutoAudioSink <TITLE>autoaudiosink</TITLE> diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index 19bc7f5d..19537ee2 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -16648,3 +16648,23 @@ <DEFAULT>0</DEFAULT> </ARG> +<ARG> +<NAME>GstAudioAmplify::amplification</NAME> +<TYPE>gfloat</TYPE> +<RANGE>>= 0</RANGE> +<FLAGS>rw</FLAGS> +<NICK>Amplification</NICK> +<BLURB>Factor of amplification.</BLURB> +<DEFAULT>1</DEFAULT> +</ARG> + +<ARG> +<NAME>GstAudioAmplify::clipping-method</NAME> +<TYPE>GstAudioPanoramaClippingMethod</TYPE> +<RANGE></RANGE> +<FLAGS>rw</FLAGS> +<NICK>Clipping method</NICK> +<BLURB>Selects how to handle values higher than the maximum.</BLURB> +<DEFAULT>Normal Clipping (default)</DEFAULT> +</ARG> + diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml index 54b4926c..f860bd2b 100644 --- a/docs/plugins/inspect/plugin-audiofx.xml +++ b/docs/plugins/inspect/plugin-audiofx.xml @@ -10,6 +10,13 @@ <origin>http://gstreamer.net/</origin> <elements> <element> + <name>audioamplify</name> + <longname>AudioAmplify</longname> + <class>Filter/Effect/Audio</class> + <description>Amplifies an audio stream by a given factor</description> + <author>Sebastian Dröge <slomo@circular-chaos.org></author> + </element> + <element> <name>audioinvert</name> <longname>AudioInvert</longname> <class>Filter/Effect/Audio</class> diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index 0b8be7a9..b3ac0b1a 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -5,7 +5,8 @@ plugin_LTLIBRARIES = libgstaudiofx.la # sources used to compile this plug-in libgstaudiofx_la_SOURCES = audiofx.c\ audiopanorama.c \ - audioinvert.c + audioinvert.c \ + audioamplify.c # flags used to compile this plugin libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \ @@ -18,4 +19,5 @@ libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) # headers we need but don't want installed noinst_HEADERS = audiopanorama.h \ - audioinvert.h + audioinvert.h \ + audioamplify.h diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c new file mode 100644 index 00000000..076aa431 --- /dev/null +++ b/gst/audiofx/audioamplify.c @@ -0,0 +1,451 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-audioamplify + * @short_description: Amplifies an audio stream with selectable clipping mode + * + * <refsect2> + * Amplifies an audio stream by a given factor and allows the selection of different clipping modes. + * The difference between the clipping modes is best evaluated by testing. + * <title>Example launch line</title> + * <para> + * <programlisting> + * gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink + * gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink + * </programlisting> + * </para> + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <gst/gst.h> +#include <gst/base/gstbasetransform.h> +#include <gst/controller/gstcontroller.h> + +#include "audioamplify.h" + +#define GST_CAT_DEFAULT gst_audio_amplify_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioAmplify", + "Filter/Effect/Audio", + "Amplifies an audio stream by a given factor", + "Sebastian Dröge <slomo@circular-chaos.org>"); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_AMPLIFICATION, + PROP_CLIPPING_METHOD +}; + +enum +{ + METHOD_CLIP = 0, + METHOD_WRAP_NEGATIVE, + METHOD_WRAP_POSITIVE, + NUM_METHODS +}; + +#define GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD (gst_audio_amplify_clipping_method_get_type ()) +static GType +gst_audio_amplify_clipping_method_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {METHOD_CLIP, "Normal Clipping (default)", "clip"}, + {METHOD_WRAP_NEGATIVE, + "Push overdriven values back from the opposite side", + "wrap-negative"}, + {METHOD_WRAP_POSITIVE, "Push overdriven values back from the same side", + "wrap-positive"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioPanoramaClippingMethod", values); + } + return gtype; +} + +static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-float, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " "width = (int) 32; " + "audio/x-raw-int, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true") + ); + +static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-float, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX], " + "endianness = (int) BYTE_ORDER, " "width = (int) 32; " + "audio/x-raw-int, " + "rate = (int) [ 1, MAX ], " + "channels = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true") + ); + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0, "audioamplify element"); + +GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstBaseTransform, + GST_TYPE_BASE_TRANSFORM, DEBUG_INIT); + +static void gst_audio_amplify_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_audio_amplify_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_amplify_set_caps (GstBaseTransform * base, + GstCaps * incaps, GstCaps * outcaps); +static GstFlowReturn gst_audio_amplify_transform_ip (GstBaseTransform * base, + GstBuffer * buf); + +static void gst_audio_amplify_transform_int_clip (GstAudioAmplify * filter, + gint16 * data, guint num_samples); +static void gst_audio_amplify_transform_int_wrap_negative (GstAudioAmplify * + filter, gint16 * data, guint num_samples); +static void gst_audio_amplify_transform_int_wrap_positive (GstAudioAmplify * + filter, gint16 * data, guint num_samples); +static void gst_audio_amplify_transform_float_clip (GstAudioAmplify * filter, + gfloat * data, guint num_samples); +static void gst_audio_amplify_transform_float_wrap_negative (GstAudioAmplify * + filter, gfloat * data, guint num_samples); +static void gst_audio_amplify_transform_float_wrap_positive (GstAudioAmplify * + filter, gfloat * data, guint num_samples); + +/* table of processing functions: [format][clipping_method] */ +static GstAudioAmplifyProcessFunc processing_functions[2][3] = { + { + (GstAudioAmplifyProcessFunc) gst_audio_amplify_transform_int_clip, + (GstAudioAmplifyProcessFunc) + gst_audio_amplify_transform_int_wrap_negative, + (GstAudioAmplifyProcessFunc) + gst_audio_amplify_transform_int_wrap_positive}, + { + (GstAudioAmplifyProcessFunc) gst_audio_amplify_transform_float_clip, + (GstAudioAmplifyProcessFunc) + gst_audio_amplify_transform_float_wrap_negative, + (GstAudioAmplifyProcessFunc) + gst_audio_amplify_transform_float_wrap_positive} +}; + +/* GObject vmethod implementations */ + +static void +gst_audio_amplify_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_template)); + gst_element_class_set_details (element_class, &element_details); +} + +static void +gst_audio_amplify_class_init (GstAudioAmplifyClass * klass) +{ + GObjectClass *gobject_class; + + gobject_class = (GObjectClass *) klass; + gobject_class->set_property = gst_audio_amplify_set_property; + gobject_class->get_property = gst_audio_amplify_get_property; + + g_object_class_install_property (gobject_class, PROP_AMPLIFICATION, + g_param_spec_float ("amplification", "Amplification", + "Factor of amplification", 0.0, G_MAXFLOAT, + 1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + /** + * GstAudioAmplify:clipping-method + * + * Clipping method: clip mode set values higher than the maximum to the + * maximum. The wrap-negative mode pushes those values back from the + * opposite side, wrap-positive pushes them back from the same side. + * + **/ + g_object_class_install_property (gobject_class, PROP_CLIPPING_METHOD, + g_param_spec_enum ("clipping-method", "Clipping method", + "Selects how to handle values higher than the maximum", + GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD, METHOD_CLIP, + G_PARAM_READWRITE)); + + GST_BASE_TRANSFORM_CLASS (klass)->set_caps = + GST_DEBUG_FUNCPTR (gst_audio_amplify_set_caps); + GST_BASE_TRANSFORM_CLASS (klass)->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_amplify_transform_ip); +} + +static void +gst_audio_amplify_init (GstAudioAmplify * filter, GstAudioAmplifyClass * klass) +{ + filter->amplification = 1.0; + filter->clipping_method = METHOD_CLIP; + filter->width = 0; + filter->format_float = FALSE; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); +} + +static gboolean +gst_audio_amplify_set_process_function (GstAudioAmplify * filter) +{ + gint format_index, method_index; + + /* set processing function */ + + format_index = (filter->format_float) ? 1 : 0; + + method_index = filter->clipping_method; + if (method_index >= NUM_METHODS || method_index < 0) + method_index = METHOD_CLIP; + + filter->process = processing_functions[format_index][method_index]; + return TRUE; +} + +static void +gst_audio_amplify_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object); + + switch (prop_id) { + case PROP_AMPLIFICATION: + filter->amplification = g_value_get_float (value); + gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), + filter->amplification == 1.0); + break; + case PROP_CLIPPING_METHOD: + filter->clipping_method = g_value_get_enum (value); + gst_audio_amplify_set_process_function (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_amplify_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object); + + switch (prop_id) { + case PROP_AMPLIFICATION: + g_value_set_float (value, filter->amplification); + break; + case PROP_CLIPPING_METHOD: + g_value_set_enum (value, filter->clipping_method); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstBaseTransform vmethod implementations */ + +static gboolean +gst_audio_amplify_set_caps (GstBaseTransform * base, GstCaps * incaps, + GstCaps * outcaps) +{ + GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base); + const GstStructure *structure; + gboolean ret; + gint width; + const gchar *fmt; + + /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */ + + structure = gst_caps_get_structure (incaps, 0); + + ret = gst_structure_get_int (structure, "width", &width); + if (!ret) + goto no_width; + filter->width = width / 8; + + + fmt = gst_structure_get_name (structure); + if (!strcmp (fmt, "audio/x-raw-int")) + filter->format_float = FALSE; + else + filter->format_float = TRUE; + + GST_DEBUG ("try to process %s input", fmt); + ret = gst_audio_amplify_set_process_function (filter); + if (!ret) + GST_WARNING ("can't process input"); + + return TRUE; + +no_width: + GST_DEBUG ("no width in caps"); + return FALSE; +} + +static void +gst_audio_amplify_transform_int_clip (GstAudioAmplify * filter, + gint16 * data, guint num_samples) +{ + gint i; + glong val; + + for (i = 0; i < num_samples; i++) { + val = (*data) * filter->amplification; + *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16); + } +} + +static void +gst_audio_amplify_transform_int_wrap_negative (GstAudioAmplify * filter, + gint16 * data, guint num_samples) +{ + gint i; + glong val; + + for (i = 0; i < num_samples; i++) { + val = (*data) * filter->amplification; + if (val > G_MAXINT16) + val = ((val - G_MININT16) & 0xffff) + G_MININT16; + else if (val < G_MININT16) + val = ((val - G_MAXINT16) & 0xffff) + G_MAXINT16; + *data++ = val; + } +} + +static void +gst_audio_amplify_transform_int_wrap_positive (GstAudioAmplify * filter, + gint16 * data, guint num_samples) +{ + gint i; + glong val; + + for (i = 0; i < num_samples; i++) { + val = (*data) * filter->amplification; + while (val > G_MAXINT16 || val < G_MININT16) { + if (val > G_MAXINT16) + val = G_MAXINT16 - (val - G_MAXINT16); + else if (val < G_MININT16) + val = G_MININT16 - (val - G_MININT16); + } + *data++ = val; + } +} + +static void +gst_audio_amplify_transform_float_clip (GstAudioAmplify * filter, + gfloat * data, guint num_samples) +{ + gint i; + gfloat val; + + for (i = 0; i < num_samples; i++) { + val = (*data) * filter->amplification; + if (val > 1.0) + val = 1.0; + else if (val < -1.0) + val = -1.0; + + *data++ = val; + } +} + +static void +gst_audio_amplify_transform_float_wrap_negative (GstAudioAmplify * filter, + gfloat * data, guint num_samples) +{ + gint i; + gfloat val; + + for (i = 0; i < num_samples; i++) { + val = (*data) * filter->amplification; + while (val > 1.0 || val < -1.0) { + if (val > 1.0) + val = -1.0 + (val - 1.0); + else if (val < -1.0) + val = 1.0 + (val + 1.0); + } + *data++ = val; + } +} + +static void +gst_audio_amplify_transform_float_wrap_positive (GstAudioAmplify * filter, + gfloat * data, guint num_samples) +{ + gint i; + gfloat val; + + for (i = 0; i < num_samples; i++) { + val = (*data) * filter->amplification; + while (val > 1.0 || val < -1.0) { + if (val > 1.0) + val = 1.0 - (val - 1.0); + else if (val < -1.0) + val = -1.0 - (val + 1.0); + } + *data++ = val; + } +} + +/* this function does the actual processing + */ +static GstFlowReturn +gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf) +{ + GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base); + guint num_samples = GST_BUFFER_SIZE (buf) / filter->width; + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} diff --git a/gst/audiofx/audioamplify.h b/gst/audiofx/audioamplify.h new file mode 100644 index 00000000..5b8de443 --- /dev/null +++ b/gst/audiofx/audioamplify.h @@ -0,0 +1,61 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net> + * + * 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_AMPLIFY_H__ +#define __GST_AUDIO_AMPLIFY_H__ + +#include <gst/gst.h> +#include <gst/base/gstbasetransform.h> + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_AMPLIFY (gst_audio_amplify_get_type()) +#define GST_AUDIO_AMPLIFY(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplify)) +#define GST_IS_AUDIO_AMPLIFY(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AMPLIFY)) +#define GST_AUDIO_AMPLIFY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplifyClass)) +#define GST_IS_AUDIO_AMPLIFY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_AMPLIFY)) +#define GST_AUDIO_AMPLIFY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplifyClass)) +typedef struct _GstAudioAmplify GstAudioAmplify; +typedef struct _GstAudioAmplifyClass GstAudioAmplifyClass; + +typedef void (*GstAudioAmplifyProcessFunc) (GstAudioAmplify *, guint8 *, guint); + +struct _GstAudioAmplify +{ + GstBaseTransform element; + + gfloat amplification; + + /* < private > */ + GstAudioAmplifyProcessFunc process; + gint clipping_method; + gint width; + gboolean format_float; +}; + +struct _GstAudioAmplifyClass +{ + GstBaseTransformClass parent; +}; + +GType gst_audio_amplify_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_AMPLIFY_H__ */ diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c index 541aa143..1ee1da66 100644 --- a/gst/audiofx/audiofx.c +++ b/gst/audiofx/audiofx.c @@ -27,6 +27,7 @@ #include "audiopanorama.h" #include "audioinvert.h" +#include "audioamplify.h" /* entry point to initialize the plug-in * initialize the plug-in itself @@ -42,7 +43,9 @@ plugin_init (GstPlugin * plugin) return (gst_element_register (plugin, "audiopanorama", GST_RANK_NONE, GST_TYPE_AUDIO_PANORAMA) && gst_element_register (plugin, "audioinvert", GST_RANK_NONE, - GST_TYPE_AUDIO_INVERT)); + GST_TYPE_AUDIO_INVERT) && + gst_element_register (plugin, "audioamplify", GST_RANK_NONE, + GST_TYPE_AUDIO_AMPLIFY)); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 9b6fac23..53121d87 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -35,6 +35,7 @@ check_PROGRAMS = \ $(check_annodex) \ elements/audiopanorama \ elements/audioinvert \ + elements/audioamplify \ elements/avimux \ elements/level \ elements/matroskamux \ diff --git a/tests/check/elements/audioamplify.c b/tests/check/elements/audioamplify.c new file mode 100644 index 00000000..a1899c3a --- /dev/null +++ b/tests/check/elements/audioamplify.c @@ -0,0 +1,469 @@ +/* GStreamer + * + * unit test for audioamplify + * + * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> + * + * Greatly based on the audiopanorama unit test + * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#include <unistd.h> + +#include <gst/base/gstbasetransform.h> +#include <gst/check/gstcheck.h> + +gboolean have_eos = FALSE; + +/* For ease of programming we use globals to keep refs for our floating + * src and sink pads we create; otherwise we always have to do get_pad, + * get_peer, and then remove references in every test function */ +GstPad *mysrcpad, *mysinkpad; + + +#define AMPLIFY_CAPS_STRING \ + "audio/x-raw-int, " \ + "channels = (int) 1, " \ + "rate = (int) 44100, " \ + "endianness = (int) BYTE_ORDER, " \ + "width = (int) 16, " \ + "depth = (int) 16, " \ + "signed = (bool) TRUE" + +static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "channels = (int) 1, " + "rate = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE") + ); +static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "channels = (int) 1, " + "rate = (int) [ 1, MAX ], " + "endianness = (int) BYTE_ORDER, " + "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE") + ); + +GstElement * +setup_amplify () +{ + GstElement *amplify; + + GST_DEBUG ("setup_amplify"); + amplify = gst_check_setup_element ("audioamplify"); + mysrcpad = gst_check_setup_src_pad (amplify, &srctemplate, NULL); + mysinkpad = gst_check_setup_sink_pad (amplify, &sinktemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); + + return amplify; +} + +void +cleanup_amplify (GstElement * amplify) +{ + GST_DEBUG ("cleanup_amplify"); + + 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 (amplify); + gst_check_teardown_sink_pad (amplify); + gst_check_teardown_element (amplify); +} + +GST_START_TEST (test_passthrough) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 *res; + + amplify = setup_amplify (); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_CAPS_STRING); + gst_buffer_set_caps (inbuffer, caps); + gst_caps_unref (caps); + ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); + + /* pushing gives away my reference ... */ + fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); + /* ... but it ends up being collected on the global buffer list */ + fail_unless_equals_int (g_list_length (buffers), 1); + fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); + + res = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + in[0], in[1], in[2], in[3], in[4], in[5], res[0], res[1], res[2], res[3], + res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), in, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_zero) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { 0, 0, 0, 0, 0, 0 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 0.0, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_050_clip) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { 12288, -8192, 128, -64, 0, -12288 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 0.5, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_200_clip) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { G_MAXINT16, -32768, 512, -256, 0, G_MININT16 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 2.0, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_050_wrap_negative) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { 12288, -8192, 128, -64, 0, -12288 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 0.5, NULL); + g_object_set (G_OBJECT (amplify), "clipping-method", 1, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_200_wrap_negative) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { -16384, -32768, 512, -256, 0, 16384 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 2.0, NULL); + g_object_set (G_OBJECT (amplify), "clipping-method", 1, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_050_wrap_positive) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { 12288, -8192, 128, -64, 0, -12288 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 0.5, NULL); + g_object_set (G_OBJECT (amplify), "clipping-method", 2, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +GST_START_TEST (test_200_wrap_positive) +{ + GstElement *amplify; + GstBuffer *inbuffer, *outbuffer; + GstCaps *caps; + gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 }; + gint16 out[6] = { 16382, -32768, 512, -256, 0, -16384 }; + gint16 *res; + + amplify = setup_amplify (); + g_object_set (G_OBJECT (amplify), "amplification", 2.0, NULL); + g_object_set (G_OBJECT (amplify), "clipping-method", 2, NULL); + fail_unless (gst_element_set_state (amplify, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "could not set to playing"); + + inbuffer = gst_buffer_new_and_alloc (12); + memcpy (GST_BUFFER_DATA (inbuffer), in, 12); + fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0); + caps = gst_caps_from_string (AMPLIFY_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 = (gint16 *) GST_BUFFER_DATA (outbuffer); + GST_INFO + ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld", + out[0], out[1], out[2], out[3], out[4], out[5], res[0], res[1], res[2], + res[3], res[4], res[5]); + fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, 12) == 0); + + /* cleanup */ + cleanup_amplify (amplify); +} + +GST_END_TEST; + +Suite * +amplify_suite (void) +{ + Suite *s = suite_create ("amplify"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + tcase_add_test (tc_chain, test_passthrough); + tcase_add_test (tc_chain, test_zero); + tcase_add_test (tc_chain, test_050_clip); + tcase_add_test (tc_chain, test_200_clip); + tcase_add_test (tc_chain, test_050_wrap_negative); + tcase_add_test (tc_chain, test_200_wrap_negative); + tcase_add_test (tc_chain, test_050_wrap_positive); + tcase_add_test (tc_chain, test_200_wrap_positive); + return s; +} + +int +main (int argc, char **argv) +{ + int nf; + + Suite *s = amplify_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; +} |