Commit 445bf71b authored by Wim Taymans's avatar Wim Taymans

port to more audio api changes

parent 77ad0a13
......@@ -112,7 +112,7 @@ static void gst_audio_amplify_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_amplify_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
static GstFlowReturn gst_audio_amplify_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
......@@ -433,7 +433,7 @@ gst_audio_amplify_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_amplify_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_amplify_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
......
......@@ -107,7 +107,7 @@ static void gst_audio_cheb_band_get_property (GObject * object,
static void gst_audio_cheb_band_finalize (GObject * object);
static gboolean gst_audio_cheb_band_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
enum
{
......@@ -641,7 +641,7 @@ gst_audio_cheb_band_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_cheb_band_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_cheb_band_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
......
......@@ -102,7 +102,7 @@ static void gst_audio_cheb_limit_get_property (GObject * object,
static void gst_audio_cheb_limit_finalize (GObject * object);
static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
enum
{
......@@ -550,7 +550,7 @@ gst_audio_cheb_limit_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_cheb_limit_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_cheb_limit_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
......
......@@ -83,7 +83,7 @@ static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
......@@ -323,7 +323,7 @@ gst_audio_dynamic_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_dynamic_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_dynamic_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
gboolean ret = TRUE;
......
......@@ -82,7 +82,7 @@ static void gst_audio_echo_get_property (GObject * object, guint prop_id,
static void gst_audio_echo_finalize (GObject * object);
static gboolean gst_audio_echo_setup (GstAudioFilter * self,
GstAudioInfo * info);
const GstAudioInfo * info);
static gboolean gst_audio_echo_stop (GstBaseTransform * base);
static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
......@@ -271,7 +271,7 @@ gst_audio_echo_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_echo_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_echo_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioEcho *self = GST_AUDIO_ECHO (base);
gboolean ret = TRUE;
......
......@@ -86,7 +86,7 @@ static void gst_audio_fir_filter_get_property (GObject * object, guint prop_id,
static void gst_audio_fir_filter_finalize (GObject * object);
static gboolean gst_audio_fir_filter_setup (GstAudioFilter * base,
GstAudioInfo * info);
const GstAudioInfo * info);
static void
......@@ -183,7 +183,7 @@ gst_audio_fir_filter_init (GstAudioFIRFilter * self)
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fir_filter_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_fir_filter_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioFIRFilter *self = GST_AUDIO_FIR_FILTER (base);
gint new_rate = GST_AUDIO_INFO_RATE (info);
......
......@@ -70,7 +70,7 @@ static gboolean gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform *
base, GstPadDirection direction, GstCaps * caps, gsize size,
GstCaps * othercaps, gsize * othersize);
static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
GstAudioInfo * info);
const GstAudioInfo * info);
static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,
GstQuery * query);
......@@ -743,7 +743,8 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
const GstAudioInfo * info)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
......
......@@ -46,7 +46,7 @@ G_DEFINE_TYPE (GstAudioFXBaseIIRFilter,
gst_audio_fx_base_iir_filter, GST_TYPE_AUDIO_FILTER);
static gboolean gst_audio_fx_base_iir_filter_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
static GstFlowReturn
gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
......@@ -236,7 +236,8 @@ gst_audio_fx_base_iir_filter_set_coefficients (GstAudioFXBaseIIRFilter * filter,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_fx_base_iir_filter_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_fx_base_iir_filter_setup (GstAudioFilter * base,
const GstAudioInfo * info)
{
GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
gboolean ret = TRUE;
......
......@@ -82,7 +82,7 @@ static void gst_audio_iir_filter_get_property (GObject * object, guint prop_id,
static void gst_audio_iir_filter_finalize (GObject * object);
static gboolean gst_audio_iir_filter_setup (GstAudioFilter * base,
GstAudioInfo * info);
const GstAudioInfo * info);
static void
gst_audio_iir_filter_class_init (GstAudioIIRFilterClass * klass)
......@@ -200,7 +200,7 @@ gst_audio_iir_filter_init (GstAudioIIRFilter * self)
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_iir_filter_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_iir_filter_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioIIRFilter *self = GST_AUDIO_IIR_FILTER (base);
gint new_rate = GST_AUDIO_INFO_RATE (info);
......
......@@ -78,7 +78,7 @@ static void gst_audio_invert_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_invert_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
static GstFlowReturn gst_audio_invert_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
......@@ -172,7 +172,7 @@ gst_audio_invert_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_invert_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_invert_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioInvert *filter = GST_AUDIO_INVERT (base);
gboolean ret = TRUE;
......
......@@ -85,7 +85,7 @@ static void gst_audio_karaoke_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_karaoke_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
static GstFlowReturn gst_audio_karaoke_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
......@@ -239,7 +239,7 @@ gst_audio_karaoke_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_karaoke_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_karaoke_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
gboolean ret = TRUE;
......
......@@ -149,7 +149,7 @@ static void gst_audio_wsincband_get_property (GObject * object, guint prop_id,
static void gst_audio_wsincband_finalize (GObject * object);
static gboolean gst_audio_wsincband_setup (GstAudioFilter * base,
GstAudioInfo * info);
const GstAudioInfo * info);
#define POW2(x) (x)*(x)
......@@ -371,7 +371,7 @@ gst_audio_wsincband_build_kernel (GstAudioWSincBand * self)
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_wsincband_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_wsincband_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base);
......
......@@ -148,7 +148,7 @@ static void gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
static void gst_audio_wsinclimit_finalize (GObject * object);
static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base,
GstAudioInfo * info);
const GstAudioInfo * info);
#define POW2(x) (x)*(x)
......@@ -306,7 +306,7 @@ gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_wsinclimit_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_audio_wsinclimit_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
......
......@@ -2,7 +2,7 @@ plugin_LTLIBRARIES = libgstauparse.la
libgstauparse_la_SOURCES = gstauparse.c
libgstauparse_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstauparse_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
libgstauparse_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR)
libgstauparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstauparse_la_LIBTOOLFLAGS = --tag=disable-static
......
......@@ -43,7 +43,7 @@ static void gst_iir_equalizer_child_proxy_interface_init (gpointer g_iface,
static void gst_iir_equalizer_finalize (GObject * object);
static gboolean gst_iir_equalizer_setup (GstAudioFilter * filter,
GstAudioInfo * info);
const GstAudioInfo * info);
static GstFlowReturn gst_iir_equalizer_transform_ip (GstBaseTransform * btrans,
GstBuffer * buf);
......@@ -848,7 +848,7 @@ gst_iir_equalizer_transform_ip (GstBaseTransform * btrans, GstBuffer * buf)
}
static gboolean
gst_iir_equalizer_setup (GstAudioFilter * audio, GstAudioInfo * info)
gst_iir_equalizer_setup (GstAudioFilter * audio, const GstAudioInfo * info)
{
GstIirEqualizer *equ = GST_IIR_EQUALIZER (audio);
......
......@@ -42,6 +42,7 @@
#include <gst/gst.h>
#include "gstgoom.h"
#include <gst/video/video.h>
#include <gst/audio/audio.h>
#include "goom.h"
#if HAVE_ORC
......@@ -82,11 +83,8 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", /* the name of the pads */
GST_PAD_SINK, /* type of the pad */
GST_PAD_ALWAYS, /* ALWAYS/SOMETIMES */
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"rate = (int) [ 8000, 96000 ], " "channels = (int) { 1, 2 }")
);
......
......@@ -2,7 +2,7 @@ plugin_LTLIBRARIES = libgstlevel.la
libgstlevel_la_SOURCES = gstlevel.c
libgstlevel_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstlevel_la_LIBADD = $(GST_BASE_LIBS) $(LIBM)
libgstlevel_la_LIBADD = $(GST_BASE_LIBS) $(LIBM) -lgstaudio-$(GST_MAJORMINOR)
libgstlevel_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstlevel_la_LIBTOOLFLAGS = --tag=disable-static
......
......@@ -9,7 +9,7 @@ libgstreplaygain_la_SOURCES = \
libgstreplaygain_la_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstreplaygain_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) -lgstpbutils-$(GST_MAJORMINOR) \
$(GST_PLUGINS_BASE_LIBS) -lgstpbutils-$(GST_MAJORMINOR) -lgstaudio-$(GST_MAJORMINOR)\
$(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstreplaygain_la_LIBTOOLFLAGS = --tag=disable-static
......
......@@ -79,6 +79,7 @@
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include "gstrganalysis.h"
#include "replaygain.h"
......@@ -103,27 +104,20 @@ enum
* audio. The used implementation has filter coefficients for the
* "usual" sample rates in the 8000 to 48000 Hz range. */
#define REPLAY_GAIN_CAPS \
"format = (string) { "GST_AUDIO_NE(F32)","GST_AUDIO_NE(S16)" }, " \
"channels = (int) { 1, 2 }, " \
"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, " \
"44100, 48000 }"
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS "; "
"audio/x-raw-int, "
"width = (int) 16, " "depth = (int) [ 1, 16 ], "
"signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS));
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (REPLAY_GAIN_CAPS));
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS "; "
"audio/x-raw-int, "
"width = (int) 16, " "depth = (int) [ 1, 16 ], "
"signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS));
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (REPLAY_GAIN_CAPS));
#define gst_rg_analysis_parent_class parent_class
G_DEFINE_TYPE (GstRgAnalysis, gst_rg_analysis, GST_TYPE_BASE_TRANSFORM);
......@@ -396,9 +390,8 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
GstCaps * out_caps)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
GstStructure *structure;
const gchar *name;
gint n_channels, sample_rate, sample_bit_size, sample_size;
GstAudioInfo info;
gint rate, channels;
g_return_val_if_fail (filter->ctx != NULL, FALSE);
......@@ -406,58 +399,42 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
"set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT,
in_caps, out_caps);
structure = gst_caps_get_structure (in_caps, 0);
name = gst_structure_get_name (structure);
if (!gst_structure_get_int (structure, "width", &sample_bit_size)
|| !gst_structure_get_int (structure, "channels", &n_channels)
|| !gst_structure_get_int (structure, "rate", &sample_rate))
if (!gst_audio_info_from_caps (&info, in_caps))
goto invalid_format;
if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate))
goto invalid_format;
rate = GST_AUDIO_INFO_RATE (&info);
if (sample_bit_size % 8 != 0)
if (!rg_analysis_set_sample_rate (filter->ctx, rate))
goto invalid_format;
sample_size = sample_bit_size / 8;
if (g_str_equal (name, "audio/x-raw-float")) {
if (sample_size != sizeof (gfloat))
goto invalid_format;
/* The depth is not variable for float formats of course. It just
* makes the transform function nice and simple if the
* rg_analysis_analyze_* functions have a common signature. */
filter->depth = sizeof (gfloat) * 8;
channels = GST_AUDIO_INFO_CHANNELS (&info);
if (n_channels == 1)
filter->analyze = rg_analysis_analyze_mono_float;
else if (n_channels == 2)
filter->analyze = rg_analysis_analyze_stereo_float;
else
goto invalid_format;
if (channels < 1 || channels > 2)
goto invalid_format;
} else if (g_str_equal (name, "audio/x-raw-int")) {
switch (GST_AUDIO_INFO_FORMAT (&info)) {
case GST_AUDIO_FORMAT_F32:
/* The depth is not variable for float formats of course. It just
* makes the transform function nice and simple if the
* rg_analysis_analyze_* functions have a common signature. */
filter->depth = sizeof (gfloat) * 8;
if (sample_size != sizeof (gint16))
goto invalid_format;
if (channels == 1)
filter->analyze = rg_analysis_analyze_mono_float;
else
filter->analyze = rg_analysis_analyze_stereo_float;
if (!gst_structure_get_int (structure, "depth", &filter->depth))
goto invalid_format;
if (filter->depth < 1 || filter->depth > 16)
goto invalid_format;
break;
case GST_AUDIO_FORMAT_S16:
filter->depth = sizeof (gint16) * 8;
if (n_channels == 1)
filter->analyze = rg_analysis_analyze_mono_int16;
else if (n_channels == 2)
filter->analyze = rg_analysis_analyze_stereo_int16;
else
if (channels == 1)
filter->analyze = rg_analysis_analyze_mono_int16;
else
filter->analyze = rg_analysis_analyze_stereo_int16;
break;
default:
goto invalid_format;
} else {
goto invalid_format;
}
return TRUE;
......
......@@ -45,6 +45,7 @@
#include <gst/gst.h>
#include <math.h>
#include <gst/audio/audio.h>
#include "gstrglimiter.h"
......@@ -58,14 +59,18 @@ enum
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, channels = (int) [1, MAX], "
"rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
"channels = (int) [1, MAX], " "rate = (int) [1, MAX]"));
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, channels = (int) [1, MAX], "
"rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
"channels = (int) [1, MAX], " "rate = (int) [1, MAX]"));
#define gst_rg_limiter_parent_class parent_class
G_DEFINE_TYPE (GstRgLimiter, gst_rg_limiter, GST_TYPE_BASE_TRANSFORM);
......
......@@ -61,6 +61,7 @@
#include <gst/gst.h>
#include <gst/pbutils/pbutils.h>
#include <gst/audio/audio.h>
#include <math.h>
#include "gstrgvolume.h"
......@@ -96,29 +97,21 @@ enum
/* Same template caps as GstVolume, for I don't like having just ANY caps. */
#define FORMAT "{ "GST_AUDIO_NE(F32)","GST_AUDIO_NE(S16)" }"
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, "
"channels = (int) [ 1, MAX ], "
"rate = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMAT ", "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
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, "
"channels = (int) [ 1, MAX ], "
"rate = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMAT ", "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
#define gst_rg_volume_parent_class parent_class
G_DEFINE_TYPE (GstRgVolume, gst_rg_volume, GST_TYPE_BIN);
......@@ -186,7 +179,7 @@ gst_rg_volume_class_init (GstRgVolumeClass * klass)
* This element internally uses a volume element, which also supports
* operating on integer audio formats. These formats do not allow exceeding
* digital full scale. If extra headroom is used, make sure that the raw
* audio data format is floating point (audio/x-raw-float). Otherwise,
* audio data format is floating point (F32). Otherwise,
* clipping distortion might be introduced as part of the volume adjustment
* itself.
*/
......
......@@ -152,7 +152,8 @@ static gboolean gst_spectrum_start (GstBaseTransform * trans);
static gboolean gst_spectrum_stop (GstBaseTransform * trans);
static GstFlowReturn gst_spectrum_transform_ip (GstBaseTransform * trans,
GstBuffer * in);
static gboolean gst_spectrum_setup (GstAudioFilter * base, GstAudioInfo * info);
static gboolean gst_spectrum_setup (GstAudioFilter * base,
const GstAudioInfo * info);
static void
gst_spectrum_class_init (GstSpectrumClass * klass)
......@@ -614,7 +615,7 @@ input_data_int16_max (const guint8 * _in, gfloat * out, guint len,
}
static gboolean
gst_spectrum_setup (GstAudioFilter * base, GstAudioInfo * info)
gst_spectrum_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstSpectrum *spectrum = GST_SPECTRUM (base);
gboolean multi_channel = spectrum->multi_channel;
......
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