port to more audio api changes

gst/audiofx/audioamplify.c

@@ -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);
 

gst/audiofx/audiochebband.c

@@ -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);
 

gst/audiofx/audiocheblimit.c

@@ -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);
 

gst/audiofx/audiodynamic.c

@@ -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;

gst/audiofx/audioecho.c

@@ -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;

gst/audiofx/audiofirfilter.c

@@ -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);

gst/audiofx/audiofxbasefirfilter.c

@@ -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);
 

gst/audiofx/audiofxbaseiirfilter.c

@@ -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;

gst/audiofx/audioiirfilter.c

@@ -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);

gst/audiofx/audioinvert.c

@@ -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;

gst/audiofx/audiokaraoke.c

@@ -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;

gst/audiofx/audiowsincband.c

@@ -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);
 

gst/audiofx/audiowsinclimit.c

@@ -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);
 

gst/auparse/Makefile.am

@@ -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
 

gst/equalizer/gstiirequalizer.c

@@ -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);
 

gst/goom/gstgoom.c

@@ -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, "
+    GST_STATIC_CAPS ("audio/x-raw, "
-        "endianness = (int) BYTE_ORDER, "
+        "format = (string) " GST_AUDIO_NE (S16) ", "
-        "signed = (boolean) TRUE, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
         "rate = (int) [ 8000, 96000 ], " "channels = (int) { 1, 2 }")
     );
 

gst/level/Makefile.am

@@ -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
 

gst/replaygain/Makefile.am

@@ -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

gst/replaygain/gstrganalysis.c

@@ -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, "
+    GST_PAD_SINK,
-        "width = (int) 32, " "endianness = (int) BYTE_ORDER, "
+    GST_PAD_ALWAYS,
-        REPLAY_GAIN_CAPS "; "
+    GST_STATIC_CAPS (REPLAY_GAIN_CAPS));
-        "audio/x-raw-int, "
-        "width = (int) 16, " "depth = (int) [ 1, 16 ], "
-        "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
-        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, "
+    GST_PAD_SRC,
-        "width = (int) 32, " "endianness = (int) BYTE_ORDER, "
+    GST_PAD_ALWAYS,
-        REPLAY_GAIN_CAPS "; "
+    GST_STATIC_CAPS (REPLAY_GAIN_CAPS));
-        "audio/x-raw-int, "
-        "width = (int) 16, " "depth = (int) [ 1, 16 ], "
-        "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
-        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;
+  GstAudioInfo info;
-  const gchar *name;
+  gint rate, channels;
-  gint n_channels, sample_rate, sample_bit_size, sample_size;
 
   g_return_val_if_fail (filter->ctx != NULL, FALSE);
 
@@ -406,57 +399,41 @@ 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);
+  if (!gst_audio_info_from_caps (&info, in_caps))
-  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))
     goto invalid_format;
 
-  if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate))
+  rate = GST_AUDIO_INFO_RATE (&info);
+
+  if (!rg_analysis_set_sample_rate (filter->ctx, rate))
     goto invalid_format;
 
-  if (sample_bit_size % 8 != 0)
+  channels = GST_AUDIO_INFO_CHANNELS (&info);
-    goto invalid_format;
-  sample_size = sample_bit_size / 8;
 
-  if (g_str_equal (name, "audio/x-raw-float")) {
+  if (channels < 1 || channels > 2)
-
-    if (sample_size != sizeof (gfloat))
     goto invalid_format;
 
+  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 (n_channels == 1)
+      if (channels == 1)
         filter->analyze = rg_analysis_analyze_mono_float;
-    else if (n_channels == 2)
+      else
         filter->analyze = rg_analysis_analyze_stereo_float;
-    else
-      goto invalid_format;
 
-  } else if (g_str_equal (name, "audio/x-raw-int")) {
+      break;
+    case GST_AUDIO_FORMAT_S16:
+      filter->depth = sizeof (gint16) * 8;
 
-    if (sample_size != sizeof (gint16))
+      if (channels == 1)
-      goto invalid_format;
-
-    if (!gst_structure_get_int (structure, "depth", &filter->depth))
-      goto invalid_format;
-    if (filter->depth < 1 || filter->depth > 16)
-      goto invalid_format;
-
-    if (n_channels == 1)
         filter->analyze = rg_analysis_analyze_mono_int16;
-    else if (n_channels == 2)
-      filter->analyze = rg_analysis_analyze_stereo_int16;
       else
-      goto invalid_format;
+        filter->analyze = rg_analysis_analyze_stereo_int16;
-
+      break;
-  } else {
+    default:
-
       goto invalid_format;
   }
 

gst/replaygain/gstrglimiter.c

@@ -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, "
+    GST_PAD_SINK,
-        "width = (int) 32, channels = (int) [1, MAX], "
+    GST_PAD_ALWAYS,
-        "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
+    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, "
+    GST_PAD_SRC,
-        "width = (int) 32, channels = (int) [1, MAX], "
+    GST_PAD_ALWAYS,
-        "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
+    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);

gst/replaygain/gstrgvolume.c

@@ -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, "
+    GST_PAD_SINK,
-        "rate = (int) [ 1, MAX ], "
+    GST_PAD_ALWAYS,
-        "channels = (int) [ 1, MAX ], "
+    GST_STATIC_CAPS ("audio/x-raw, "
-        "endianness = (int) BYTE_ORDER, "
+        "format = (string) " FORMAT ", "
-        "width = (int) 32; "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
-        "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"));
 
 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
+    GST_PAD_SRC,
-        "rate = (int) [ 1, MAX ], "
+    GST_PAD_ALWAYS,
-        "channels = (int) [ 1, MAX ], "
+    GST_STATIC_CAPS ("audio/x-raw, "
-        "endianness = (int) BYTE_ORDER, "
+        "format = (string) " FORMAT ", "
-        "width = (int) 32; "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
-        "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"));
 
 #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.
    */

gst/spectrum/gstspectrum.c

@@ -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;