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gstreamer/ext/bs2b/gstbs2b.c
Sebastian Dröge 80e02cbdf9 Constify some static arrays everywhere
2015-01-21 10:18:50 +01:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2003> David Schleef <ds@schleef.org>
* Copyright (C) <2011,2014> Christoph Reiter <reiter.christoph@gmail.com>
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* SECTION:element-bs2b
*
* Improve headphone listening of stereo audio records using the bs2b library.
* It does so by mixing the left and right channel in a way that simulates
* a stereo speaker setup while using headphones.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 audiotestsrc ! "audio/x-raw,channel-mask=(bitmask)0x1" ! interleave name=i ! bs2b ! autoaudiosink audiotestsrc freq=330 ! "audio/x-raw,channel-mask=(bitmask)0x2" ! i.
* ]| Play two independent sine test sources and crossfeed them.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include "gstbs2b.h"
#define GST_BS2B_DP_LOCK(obj) g_mutex_lock (&obj->bs2b_lock)
#define GST_BS2B_DP_UNLOCK(obj) g_mutex_unlock (&obj->bs2b_lock)
#define SUPPORTED_FORMAT \
"(string) { S8, U8, S16LE, S16BE, U16LE, U16BE, S32LE, S32BE, U32LE, " \
"U32BE, S24LE, S24BE, U24LE, U24BE, F32LE, F32BE, F64LE, F64BE }"
#define SUPPORTED_RATE \
"(int) [ " G_STRINGIFY (BS2B_MINSRATE) ", " G_STRINGIFY (BS2B_MAXSRATE) " ]"
#define FRONT_L_FRONT_R "(bitmask) 0x3"
#define PAD_CAPS \
"audio/x-raw, " \
"format = " SUPPORTED_FORMAT ", " \
"rate = " SUPPORTED_RATE ", " \
"channels = (int) 2, " \
"channel-mask = " FRONT_L_FRONT_R ", " \
"layout = (string) interleaved" \
"; " \
"audio/x-raw, " \
"channels = (int) 1" \
enum
{
PROP_FCUT = 1,
PROP_FEED,
PROP_LAST,
};
static GParamSpec *properties[PROP_LAST];
typedef struct
{
const gchar *name;
const gchar *desc;
gint preset;
} GstBs2bPreset;
static const GstBs2bPreset presets[3] = {
{
"default",
"Closest to virtual speaker placement (30°, 3 meter) [700Hz, 4.5dB]",
BS2B_DEFAULT_CLEVEL},
{
"cmoy",
"Close to Chu Moy's crossfeeder (popular) [700Hz, 6.0dB]",
BS2B_CMOY_CLEVEL},
{
"jmeier",
"Close to Jan Meier's CORDA amplifiers (little change) [650Hz, 9.0dB]",
BS2B_JMEIER_CLEVEL}
};
static void gst_preset_interface_init (gpointer g_iface, gpointer iface_data);
G_DEFINE_TYPE_WITH_CODE (GstBs2b, gst_bs2b, GST_TYPE_AUDIO_FILTER,
G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, gst_preset_interface_init));
static void gst_bs2b_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_bs2b_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_bs2b_finalize (GObject * object);
static GstFlowReturn gst_bs2b_transform_inplace (GstBaseTransform *
base_transform, GstBuffer * buffer);
static gboolean gst_bs2b_setup (GstAudioFilter * self,
const GstAudioInfo * audio_info);
static gchar **
gst_bs2b_get_preset_names (GstPreset * preset)
{
gchar **names;
gint i;
names = g_new (gchar *, 1 + G_N_ELEMENTS (presets));
for (i = 0; i < G_N_ELEMENTS (presets); i++) {
names[i] = g_strdup (presets[i].name);
}
names[i] = NULL;
return names;
}
static gchar **
gst_bs2b_get_property_names (GstPreset * preset)
{
gchar **names = g_new (gchar *, 3);
names[0] = g_strdup ("fcut");
names[1] = g_strdup ("feed");
names[2] = NULL;
return names;
}
static gboolean
gst_bs2b_load_preset (GstPreset * preset, const gchar * name)
{
GstBs2b *element = GST_BS2B (preset);
GObject *object = (GObject *) preset;
gint i;
for (i = 0; i < G_N_ELEMENTS (presets); i++) {
if (!g_strcmp0 (presets[i].name, name)) {
bs2b_set_level (element->bs2bdp, presets[i].preset);
g_object_notify_by_pspec (object, properties[PROP_FCUT]);
g_object_notify_by_pspec (object, properties[PROP_FEED]);
return TRUE;
}
}
return FALSE;
}
static gboolean
gst_bs2b_get_meta (GstPreset * preset, const gchar * name,
const gchar * tag, gchar ** value)
{
if (!g_strcmp0 (tag, "comment")) {
gint i;
for (i = 0; i < G_N_ELEMENTS (presets); i++) {
if (!g_strcmp0 (presets[i].name, name)) {
*value = g_strdup (presets[i].desc);
return TRUE;
}
}
}
*value = NULL;
return FALSE;
}
static void
gst_preset_interface_init (gpointer g_iface, gpointer iface_data)
{
GstPresetInterface *iface = g_iface;
iface->get_preset_names = gst_bs2b_get_preset_names;
iface->get_property_names = gst_bs2b_get_property_names;
iface->load_preset = gst_bs2b_load_preset;
iface->save_preset = NULL;
iface->rename_preset = NULL;
iface->delete_preset = NULL;
iface->get_meta = gst_bs2b_get_meta;
iface->set_meta = NULL;
}
static void
gst_bs2b_class_init (GstBs2bClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (klass);
GstCaps *caps;
gobject_class->set_property = gst_bs2b_set_property;
gobject_class->get_property = gst_bs2b_get_property;
gobject_class->finalize = gst_bs2b_finalize;
trans_class->transform_ip = gst_bs2b_transform_inplace;
trans_class->transform_ip_on_passthrough = FALSE;
filter_class->setup = gst_bs2b_setup;
properties[PROP_FCUT] = g_param_spec_int ("fcut", "Frequency cut",
"Low-pass filter cut frequency (Hz)",
BS2B_MINFCUT, BS2B_MAXFCUT, BS2B_DEFAULT_CLEVEL & 0xFFFF,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS);
properties[PROP_FEED] =
g_param_spec_int ("feed", "Feed level", "Feed Level (dB/10)",
BS2B_MINFEED, BS2B_MAXFEED, BS2B_DEFAULT_CLEVEL >> 16,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS);
g_object_class_install_properties (gobject_class, PROP_LAST, properties);
gst_element_class_set_metadata (element_class,
"Crossfeed effect",
"Filter/Effect/Audio",
"Improve headphone listening of stereo audio records using the bs2b "
"library.", "Christoph Reiter <reiter.christoph@gmail.com>");
caps = gst_caps_from_string (PAD_CAPS);
gst_audio_filter_class_add_pad_templates (filter_class, caps);
gst_caps_unref (caps);
}
static void
gst_bs2b_init (GstBs2b * element)
{
g_mutex_init (&element->bs2b_lock);
element->bs2bdp = bs2b_open ();
}
static gboolean
gst_bs2b_setup (GstAudioFilter * filter, const GstAudioInfo * audio_info)
{
GstBaseTransform *base_transform = GST_BASE_TRANSFORM (filter);
GstBs2b *element = GST_BS2B (filter);
gint channels = GST_AUDIO_INFO_CHANNELS (audio_info);
element->func = NULL;
if (channels == 1) {
gst_base_transform_set_passthrough (base_transform, TRUE);
return TRUE;
}
g_assert (channels == 2);
gst_base_transform_set_passthrough (base_transform, FALSE);
switch (GST_AUDIO_INFO_FORMAT (audio_info)) {
case GST_AUDIO_FORMAT_S8:
element->func = &bs2b_cross_feed_s8;
break;
case GST_AUDIO_FORMAT_U8:
element->func = &bs2b_cross_feed_u8;
break;
case GST_AUDIO_FORMAT_S16BE:
element->func = &bs2b_cross_feed_s16be;
break;
case GST_AUDIO_FORMAT_S16LE:
element->func = &bs2b_cross_feed_s16le;
break;
case GST_AUDIO_FORMAT_U16BE:
element->func = &bs2b_cross_feed_u16be;
break;
case GST_AUDIO_FORMAT_U16LE:
element->func = &bs2b_cross_feed_u16le;
break;
case GST_AUDIO_FORMAT_S24BE:
element->func = &bs2b_cross_feed_s24be;
break;
case GST_AUDIO_FORMAT_S24LE:
element->func = &bs2b_cross_feed_s24le;
break;
case GST_AUDIO_FORMAT_U24BE:
element->func = &bs2b_cross_feed_u24be;
break;
case GST_AUDIO_FORMAT_U24LE:
element->func = &bs2b_cross_feed_u24le;
break;
case GST_AUDIO_FORMAT_S32BE:
element->func = &bs2b_cross_feed_s32be;
break;
case GST_AUDIO_FORMAT_S32LE:
element->func = &bs2b_cross_feed_s32le;
break;
case GST_AUDIO_FORMAT_U32BE:
element->func = &bs2b_cross_feed_u32be;
break;
case GST_AUDIO_FORMAT_U32LE:
element->func = &bs2b_cross_feed_u32le;
break;
case GST_AUDIO_FORMAT_F32BE:
element->func = &bs2b_cross_feed_fbe;
break;
case GST_AUDIO_FORMAT_F32LE:
element->func = &bs2b_cross_feed_fle;
break;
case GST_AUDIO_FORMAT_F64BE:
element->func = &bs2b_cross_feed_dbe;
break;
case GST_AUDIO_FORMAT_F64LE:
element->func = &bs2b_cross_feed_dle;
break;
default:
return FALSE;
}
g_assert (element->func);
element->bytes_per_sample =
(GST_AUDIO_INFO_WIDTH (audio_info) * channels) / 8;
GST_BS2B_DP_LOCK (element);
bs2b_set_srate (element->bs2bdp, GST_AUDIO_INFO_RATE (audio_info));
GST_BS2B_DP_UNLOCK (element);
return TRUE;
}
static void
gst_bs2b_finalize (GObject * object)
{
GstBs2b *element = GST_BS2B (object);
bs2b_close (element->bs2bdp);
element->bs2bdp = NULL;
G_OBJECT_CLASS (gst_bs2b_parent_class)->finalize (object);
}
static GstFlowReturn
gst_bs2b_transform_inplace (GstBaseTransform * base_transform,
GstBuffer * buffer)
{
GstBs2b *element = GST_BS2B (base_transform);
GstMapInfo map_info;
if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ | GST_MAP_WRITE))
return GST_FLOW_ERROR;
GST_BS2B_DP_LOCK (element);
if (GST_BUFFER_IS_DISCONT (buffer))
bs2b_clear (element->bs2bdp);
element->func (element->bs2bdp, map_info.data,
map_info.size / element->bytes_per_sample);
GST_BS2B_DP_UNLOCK (element);
gst_buffer_unmap (buffer, &map_info);
return GST_FLOW_OK;
}
static void
gst_bs2b_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBs2b *element = GST_BS2B (object);
switch (prop_id) {
case PROP_FCUT:
GST_BS2B_DP_LOCK (element);
bs2b_set_level_fcut (element->bs2bdp, g_value_get_int (value));
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
break;
case PROP_FEED:
GST_BS2B_DP_LOCK (element);
bs2b_set_level_feed (element->bs2bdp, g_value_get_int (value));
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_bs2b_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBs2b *element = GST_BS2B (object);
switch (prop_id) {
case PROP_FCUT:
GST_BS2B_DP_LOCK (element);
g_value_set_int (value, bs2b_get_level_fcut (element->bs2bdp));
GST_BS2B_DP_UNLOCK (element);
break;
case PROP_FEED:
GST_BS2B_DP_LOCK (element);
g_value_set_int (value, bs2b_get_level_feed (element->bs2bdp));
GST_BS2B_DP_UNLOCK (element);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "bs2b", GST_RANK_NONE, GST_TYPE_BS2B);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
bs2b,
"Improve headphone listening of stereo audio records"
"using the bs2b library.",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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