-good: port to new audio caps

diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h
index 9724cc0..2b5dbe8 100644 (file)
--- a/ext/jack/gstjack.h
+++ b/ext/jack/gstjack.h
@@ -50,9 +50,9 @@ typedef jack_default_audio_sample_t sample_t;
 #define GST_TYPE_JACK_CLIENT  (gst_jack_client_get_type ())
 
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define GST_JACK_FORMAT_STR "F32_LE"
+#define GST_JACK_FORMAT_STR "F32LE"
 #else
-#define GST_JACK_FORMAT_STR "F32_BE"
+#define GST_JACK_FORMAT_STR "F32BE"
 #endif
 
 GType gst_jack_client_get_type(void);

diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c
index dd89f99..b4c8a54 100644 (file)
--- a/ext/pulse/pulsesink.c
+++ b/ext/pulse/pulsesink.c
@@ -1717,11 +1717,11 @@ static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
     GstStateChange transition);
 
 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define FORMATS   "{ S16_LE, S16_BE, F32_LE, F32_BE, S32_LE, S32_BE, " \
-                     "S24_3LE, S24_3BE, S24_LE, S24_BE, S8 }"
+# define FORMATS   "{ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, " \
+                     "S24LE, S24BE, S24_32LE, S24_32BE, S8 }"
 #else
-# define FORMATS   "{ S16_BE, S16_LE, F32_BE, F32_LE, S32_BE, S32_LE, " \
-                     "S24_3BE, S24_3LE, S24_BE, S24_LE, S8 }"
+# define FORMATS   "{ S16BE, S16LE, F32BE, F32LE, S32BE, S32LE, " \
+                     "S24BE, S24LE, S24_32BE, S24_32LE, S8 }"
 #endif
 
 static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",

diff --git a/ext/pulse/pulsesrc.c b/ext/pulse/pulsesrc.c
index e0c1134..112d41b 100644 (file)
--- a/ext/pulse/pulsesrc.c
+++ b/ext/pulse/pulsesrc.c
@@ -99,9 +99,9 @@ static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
     element, GstStateChange transition);
 
 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define FORMATS "{ S16_LE, S16_BE, F32_LE, F32_BE, S32_LE, S32_BE, U8 }"
+# define FORMATS "{ S16LE, S16BE, F32LE, F32BE, S32LE, S32BE, U8 }"
 #else
-# define FORMATS "{ S16_BE, S16_LE, F32_BE, F32_LE, S32_BE, S32_LE, U8 }"
+# define FORMATS "{ S16BE, S16LE, F32BE, F32LE, S32BE, S32LE, U8 }"
 #endif
 
 static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
@@ -1084,7 +1084,7 @@ gst_pulsesrc_success_cb (pa_stream * s, int success, void *userdata)
 {
   GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (userdata);
 
-  pulsesrc->operation_success = ! !success;
+  pulsesrc->operation_success = !!success;
   pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
 }
 

diff --git a/ext/pulse/pulseutil.c b/ext/pulse/pulseutil.c
index 196e39c..171ef4c 100644 (file)
--- a/ext/pulse/pulseutil.c
+++ b/ext/pulse/pulseutil.c
@@ -98,34 +98,34 @@ gst_pulse_fill_sample_spec (GstRingBufferSpec * spec, pa_sample_spec * ss)
       case GST_AUDIO_FORMAT_U8:
         ss->format = PA_SAMPLE_U8;
         break;
-      case GST_AUDIO_FORMAT_S16_LE:
+      case GST_AUDIO_FORMAT_S16LE:
         ss->format = PA_SAMPLE_S16LE;
         break;
-      case GST_AUDIO_FORMAT_S16_BE:
+      case GST_AUDIO_FORMAT_S16BE:
         ss->format = PA_SAMPLE_S16BE;
         break;
-      case GST_AUDIO_FORMAT_F32_LE:
+      case GST_AUDIO_FORMAT_F32LE:
         ss->format = PA_SAMPLE_FLOAT32LE;
         break;
-      case GST_AUDIO_FORMAT_F32_BE:
+      case GST_AUDIO_FORMAT_F32BE:
         ss->format = PA_SAMPLE_FLOAT32BE;
         break;
-      case GST_AUDIO_FORMAT_S32_LE:
+      case GST_AUDIO_FORMAT_S32LE:
         ss->format = PA_SAMPLE_S32LE;
         break;
-      case GST_AUDIO_FORMAT_S32_BE:
+      case GST_AUDIO_FORMAT_S32BE:
         ss->format = PA_SAMPLE_S32BE;
         break;
-      case GST_AUDIO_FORMAT_S24_3LE:
+      case GST_AUDIO_FORMAT_S24LE:
         ss->format = PA_SAMPLE_S24LE;
         break;
-      case GST_AUDIO_FORMAT_S24_3BE:
+      case GST_AUDIO_FORMAT_S24BE:
         ss->format = PA_SAMPLE_S24BE;
         break;
-      case GST_AUDIO_FORMAT_S24_LE:
+      case GST_AUDIO_FORMAT_S24_32LE:
         ss->format = PA_SAMPLE_S24_32LE;
         break;
-      case GST_AUDIO_FORMAT_S24_BE:
+      case GST_AUDIO_FORMAT_S24_32BE:
         ss->format = PA_SAMPLE_S24_32BE;
         break;
       default:

diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c
index aa3bc84..9c43f60 100644 (file)
--- a/gst/audiofx/audiopanorama.c
+++ b/gst/audiofx/audiopanorama.c
@@ -92,29 +92,17 @@ gst_audio_panorama_method_get_type (void)
 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, 2 ], "
-        "endianness = (int) BYTE_ORDER, " "width = (int) 32; "
-        "audio/x-raw-int, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, 2 ], "
-        "endianness = (int) BYTE_ORDER, "
-        "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { " GST_AUDIO_NE (S32) ", " GST_AUDIO_NE (S16) "}, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );
 
 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) 2, "
-        "endianness = (int) BYTE_ORDER, " "width = (int) 32; "
-        "audio/x-raw-int, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) 2, "
-        "endianness = (int) BYTE_ORDER, "
-        "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { " GST_AUDIO_NE (S32) ", " GST_AUDIO_NE (S16) "}, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) 2")
     );
 
 G_DEFINE_TYPE (GstAudioPanorama, gst_audio_panorama, GST_TYPE_BASE_TRANSFORM);
@@ -237,27 +225,27 @@ gst_audio_panorama_init (GstAudioPanorama * filter)
 
   filter->panorama = 0;
   filter->method = METHOD_PSYCHOACOUSTIC;
-  filter->width = 0;
-  filter->channels = 0;
-  filter->format_float = FALSE;
+  gst_audio_info_init (&filter->info);
   filter->process = NULL;
 
   gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
 }
 
 static gboolean
-gst_audio_panorama_set_process_function (GstAudioPanorama * filter)
+gst_audio_panorama_set_process_function (GstAudioPanorama * filter,
+    GstAudioInfo * info)
 {
   gint channel_index, format_index, method_index;
+  const GstAudioFormatInfo *finfo = info->finfo;
 
   /* set processing function */
-  channel_index = filter->channels - 1;
+  channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1;
   if (channel_index > 1 || channel_index < 0) {
     filter->process = NULL;
     return FALSE;
   }
 
-  format_index = (filter->format_float) ? 1 : 0;
+  format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0;
 
   method_index = filter->method;
   if (method_index >= NUM_METHODS || method_index < 0)
@@ -280,7 +268,7 @@ gst_audio_panorama_set_property (GObject * object, guint prop_id,
       break;
     case PROP_METHOD:
       filter->method = g_value_get_enum (value);
-      gst_audio_panorama_set_process_function (filter);
+      gst_audio_panorama_set_process_function (filter, &filter->info);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -356,44 +344,27 @@ gst_audio_panorama_set_caps (GstBaseTransform * base, GstCaps * incaps,
     GstCaps * outcaps)
 {
   GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
-  const GstStructure *structure;
-  gboolean ret;
-  gint width;
-  const gchar *fmt;
+  GstAudioInfo info;
 
   /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
+  if (!gst_audio_info_from_caps (&info, incaps))
+    goto no_format;
 
-  structure = gst_caps_get_structure (incaps, 0);
-  ret = gst_structure_get_int (structure, "channels", &filter->channels);
-  if (!ret)
-    goto no_channels;
-
-  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 with %d channels", fmt, filter->channels);
+  GST_DEBUG ("try to process %d input with %d channels",
+      GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info));
 
-  ret = gst_audio_panorama_set_process_function (filter);
+  if (!gst_audio_panorama_set_process_function (filter, &info))
+    goto no_format;
 
-  if (!ret)
-    GST_WARNING ("can't process input with %d channels", filter->channels);
+  filter->info = info;
 
-  return ret;
+  return TRUE;
 
-no_channels:
-  GST_DEBUG ("no channels in caps");
-  return ret;
-no_width:
-  GST_DEBUG ("no width in caps");
-  return ret;
+no_format:
+  {
+    GST_DEBUG ("invalid caps");
+    return FALSE;
+  }
 }
 
 /* psychoacoustic processing functions */
@@ -672,7 +643,7 @@ gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
     memset (outdata, 0, outsize);
   } else {
-    guint num_samples = outsize / (2 * filter->width);
+    guint num_samples = outsize / GST_AUDIO_INFO_BPF (&filter->info);
 
     filter->process (filter, indata, outdata, num_samples);
   }

diff --git a/gst/audiofx/audiopanorama.h b/gst/audiofx/audiopanorama.h
index b3cd183..e445f4d 100644 (file)
--- a/gst/audiofx/audiopanorama.h
+++ b/gst/audiofx/audiopanorama.h
@@ -22,6 +22,7 @@
 #define __GST_AUDIO_PANORAMA_H__
 
 #include <gst/gst.h>
+#include <gst/audio/audio.h>
 #include <gst/base/gstbasetransform.h>
 
 G_BEGIN_DECLS
@@ -42,12 +43,11 @@ struct _GstAudioPanorama {
   GstBaseTransform element;
 
   gfloat panorama;
-  
+
   /* < private > */
   GstAudioPanoramaProcessFunc process;
-  gint channels;
-  gboolean format_float;
-  gint width;
+
+  GstAudioInfo info;
   gint method;
 };
 

diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c
index 054a640..bf7b639 100644 (file)
--- a/gst/auparse/gstauparse.c
+++ b/gst/auparse/gstauparse.c
@@ -45,9 +45,9 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
 
 #define GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS \
     "audio/x-raw, "                         \
-    "format= (string) { S8, S16_LE, S16_BE, S24_3LE, S24_3BE, "  \
-                       "S32_LE, S32_BE, F32_LE, F32_BE, "        \
-                       "F64_LE, F64_BE }, " \
+    "format= (string) { S8, S16LE, S16BE, S24LE, S24BE, "  \
+                       "S32LE, S32BE, F32LE, F32BE, "        \
+                       "F64LE, F64BE }, " \
     "rate = (int) [ 8000, 192000 ], "       \
     "channels = (int) [ 1, 2 ]"
 
@@ -257,38 +257,38 @@ gst_au_parse_parse_header (GstAuParse * auparse)
       break;
     case 3:                    /* 16-bit linear PCM */
       if (endianness == G_LITTLE_ENDIAN)
-        format = GST_AUDIO_FORMAT_S16_LE;
+        format = GST_AUDIO_FORMAT_S16LE;
       else
-        format = GST_AUDIO_FORMAT_S16_BE;
+        format = GST_AUDIO_FORMAT_S16BE;
       auparse->sample_size = auparse->channels * 2;
       break;
     case 4:                    /* 24-bit linear PCM */
       if (endianness == G_LITTLE_ENDIAN)
-        format = GST_AUDIO_FORMAT_S24_3LE;
+        format = GST_AUDIO_FORMAT_S24LE;
       else
-        format = GST_AUDIO_FORMAT_S24_3BE;
+        format = GST_AUDIO_FORMAT_S24BE;
       auparse->sample_size = auparse->channels * 3;
       break;
     case 5:                    /* 32-bit linear PCM */
       if (endianness == G_LITTLE_ENDIAN)
-        format = GST_AUDIO_FORMAT_S32_LE;
+        format = GST_AUDIO_FORMAT_S32LE;
       else
-        format = GST_AUDIO_FORMAT_S32_BE;
+        format = GST_AUDIO_FORMAT_S32BE;
       auparse->sample_size = auparse->channels * 4;
       break;
 
     case 6:                    /* 32-bit IEEE floating point */
       if (endianness == G_LITTLE_ENDIAN)
-        format = GST_AUDIO_FORMAT_F32_LE;
+        format = GST_AUDIO_FORMAT_F32LE;
       else
-        format = GST_AUDIO_FORMAT_F32_BE;
+        format = GST_AUDIO_FORMAT_F32BE;
       auparse->sample_size = auparse->channels * 4;
       break;
     case 7:                    /* 64-bit IEEE floating point */
       if (endianness == G_LITTLE_ENDIAN)
-        format = GST_AUDIO_FORMAT_F64_LE;
+        format = GST_AUDIO_FORMAT_F64LE;
       else
-        format = GST_AUDIO_FORMAT_F64_BE;
+        format = GST_AUDIO_FORMAT_F64BE;
       auparse->sample_size = auparse->channels * 8;
       break;
 

diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c
index 514257e..13e0fff 100644 (file)
--- a/gst/avi/gstavimux.c
+++ b/gst/avi/gstavimux.c
@@ -38,7 +38,7 @@
  * ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' \
  * ! queue ! mux. \
  * audiotestsrc num-buffers=440 ! audioconvert \
- * ! 'audio/x-raw-int,rate=44100,channels=2' ! queue ! mux. \
+ * ! 'audio/x-raw,rate=44100,channels=2' ! queue ! mux. \
  * avimux name=mux ! filesink location=test.avi
  * ]| This will create an .AVI file containing an uncompressed video stream
  * with a test picture and an uncompressed audio stream containing a
@@ -47,7 +47,7 @@
  * gst-launch videotestsrc num-buffers=250 \
  * ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' \
  * ! xvidenc ! queue ! mux. \
- * audiotestsrc num-buffers=440 ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=2' \
+ * audiotestsrc num-buffers=440 ! audioconvert ! 'audio/x-raw,rate=44100,channels=2' \
  * ! lame ! queue ! mux. \
  * avimux name=mux ! filesink location=test.avi
  * ]| This will create an .AVI file containing the same test video and sound
@@ -66,6 +66,7 @@
 #include <string.h>
 
 #include <gst/video/video.h>
+#include <gst/audio/audio.h>
 #include <gst/base/gstbytewriter.h>
 
 #include "gstavimux.h"
@@ -160,11 +161,8 @@ static GstStaticPadTemplate audio_sink_factory =
     GST_STATIC_PAD_TEMPLATE ("audio_%d",
     GST_PAD_SINK,
     GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) LITTLE_ENDIAN, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) { 8, 16 }, "
-        "depth = (int) { 8, 16 }, "
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { U8, S16LE }, "
         "rate = (int) [ 1000, 96000 ], "
         "channels = (int) [ 1, 2 ]; "
         "audio/mpeg, "
@@ -758,36 +756,27 @@ gst_avi_mux_audsink_set_caps (GstPad * pad, GstCaps * vscaps)
     avimux->codec_data_size += gst_buffer_get_size (avipad->auds_codec_data);
   }
 
-  if (!strcmp (mimetype, "audio/x-raw-int")) {
-    gint width, depth;
-    gboolean signedness;
-
-    avipad->auds.format = GST_RIFF_WAVE_FORMAT_PCM;
-
-    if (!gst_structure_get_int (structure, "width", &width) ||
-        !gst_structure_get_int (structure, "depth", &depth) ||
-        !gst_structure_get_boolean (structure, "signed", &signedness)) {
-      GST_DEBUG_OBJECT (avimux,
-          "broken caps, width/depth/signed field missing");
-      goto refuse_caps;
-    }
+  if (!strcmp (mimetype, "audio/x-raw")) {
+    const gchar *format;
+    GstAudioFormat fmt;
 
-    /* no clear place to put different values for these while keeping to spec */
-    if (width != depth) {
-      GST_DEBUG_OBJECT (avimux, "width must be same as depth!");
-      goto refuse_caps;
-    }
+    format = gst_structure_get_string (structure, "format");
+    fmt = gst_audio_format_from_string (format);
 
-    /* because that's the way the caps will be recreated from riff data */
-    if ((width == 8 && signedness) || (width == 16 && !signedness)) {
-      GST_DEBUG_OBJECT (avimux,
-          "8-bit PCM must be unsigned, 16-bit PCM signed");
-      goto refuse_caps;
+    switch (fmt) {
+      case GST_AUDIO_FORMAT_U8:
+        avipad->auds.blockalign = 8;
+        avipad->auds.size = 8;
+        break;
+      case GST_AUDIO_FORMAT_S16:
+        avipad->auds.blockalign = 16;
+        avipad->auds.size = 16;
+        break;
+      default:
+        goto refuse_caps;
     }
 
-    avipad->auds.blockalign = width;
-    avipad->auds.size = (width == 8) ? 8 : depth;
-
+    avipad->auds.format = GST_RIFF_WAVE_FORMAT_PCM;
     /* set some more info straight */
     avipad->auds.blockalign /= 8;
     avipad->auds.blockalign *= avipad->auds.channels;

diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c
index 9dc9fba..3e210f7 100644 (file)
--- a/gst/isomp4/gstqtmux.c
+++ b/gst/isomp4/gstqtmux.c
@@ -97,7 +97,7 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch v4l2src num-buffers=500 ! video/x-raw-yuv,width=320,height=240 ! ffmpegcolorspace ! qtmux ! filesink location=video.mov
+ * gst-launch v4l2src num-buffers=500 ! video/x-raw,width=320,height=240 ! ffmpegcolorspace ! qtmux ! filesink location=video.mov
  * ]|
  * Records a video stream captured from a v4l2 device and muxes it into a qt file.
  * </refsect2>

diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
index 057d03d..d81c2dc 100644 (file)
--- a/gst/isomp4/qtdemux.c
+++ b/gst/isomp4/qtdemux.c
@@ -5256,7 +5256,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl)
   /* sync sample atom */
   stream->stps_present = FALSE;
   if ((stream->stss_present =
-          ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss,
+          !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stss,
               &stream->stss) ? TRUE : FALSE) == TRUE) {
     /* copy atom data into a new buffer for later use */
     stream->stss.data = g_memdup (stream->stss.data, stream->stss.size);
@@ -5274,7 +5274,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl)
 
     /* partial sync sample atom */
     if ((stream->stps_present =
-            ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps,
+            !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_stps,
                 &stream->stps) ? TRUE : FALSE) == TRUE) {
       /* copy atom data into a new buffer for later use */
       stream->stps.data = g_memdup (stream->stps.data, stream->stps.size);
@@ -5393,7 +5393,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl)
 
   /* composition time-to-sample */
   if ((stream->ctts_present =
-          ! !qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts,
+          !!qtdemux_tree_get_child_by_type_full (stbl, FOURCC_ctts,
               &stream->ctts) ? TRUE : FALSE) == TRUE) {
     /* copy atom data into a new buffer for later use */
     stream->ctts.data = g_memdup (stream->ctts.data, stream->ctts.size);
@@ -9382,24 +9382,24 @@ qtdemux_audio_caps (GstQTDemux * qtdemux, QtDemuxStream * stream,
     case GST_MAKE_FOURCC ('f', 'l', '6', '4'):
       _codec ("Raw 64-bit floating-point audio");
       caps = gst_caps_new_simple ("audio/x-raw",
-          "format", G_TYPE_STRING, "F64_BE", NULL);
+          "format", G_TYPE_STRING, "F64BE", NULL);
       break;
     case GST_MAKE_FOURCC ('f', 'l', '3', '2'):
       _codec ("Raw 32-bit floating-point audio");
       caps = gst_caps_new_simple ("audio/x-raw",
-          "format", G_TYPE_STRING, "F32_BE", NULL);
+          "format", G_TYPE_STRING, "F32BE", NULL);
       break;
     case FOURCC_in24:
       _codec ("Raw 24-bit PCM audio");
       /* we assume BIG ENDIAN, an enda box will tell us to change this to little
        * endian later */
       caps = gst_caps_new_simple ("audio/x-raw",
-          "format", G_TYPE_STRING, "S24_3BE", NULL);
+          "format", G_TYPE_STRING, "S24BE", NULL);
       break;
     case GST_MAKE_FOURCC ('i', 'n', '3', '2'):
       _codec ("Raw 32-bit PCM audio");
       caps = gst_caps_new_simple ("audio/x-raw",
-          "format", G_TYPE_STRING, "S32_BE", NULL);
+          "format", G_TYPE_STRING, "S32BE", NULL);
       break;
     case GST_MAKE_FOURCC ('u', 'l', 'a', 'w'):
       _codec ("Mu-law audio");

diff --git a/gst/law/alaw.c b/gst/law/alaw.c
index 1e7cc78..9e34205 100644 (file)
--- a/gst/law/alaw.c
+++ b/gst/law/alaw.c
@@ -24,9 +24,9 @@
 #include "alaw-decode.h"
 
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define INT_FORMAT "S16_LE"
+#define INT_FORMAT "S16LE"
 #else
-#define INT_FORMAT "S16_BE"
+#define INT_FORMAT "S16BE"
 #endif
 
 GstStaticPadTemplate alaw_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src",

diff --git a/gst/law/mulaw-decode.c b/gst/law/mulaw-decode.c
index 15bbcd8..e9bbdf2 100644 (file)
--- a/gst/law/mulaw-decode.c
+++ b/gst/law/mulaw-decode.c
@@ -31,9 +31,9 @@
 #include "mulaw-conversion.h"
 
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define INT_FORMAT "S16_LE"
+#define INT_FORMAT "S16LE"
 #else
-#define INT_FORMAT "S16_BE"
+#define INT_FORMAT "S16BE"
 #endif
 
 extern GstStaticPadTemplate mulaw_dec_src_factory;

diff --git a/gst/law/mulaw.c b/gst/law/mulaw.c
index 1784084..ba310db 100644 (file)
--- a/gst/law/mulaw.c
+++ b/gst/law/mulaw.c
@@ -23,9 +23,9 @@
 #include "mulaw-decode.h"
 
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define INT_FORMAT "S16_LE"
+#define INT_FORMAT "S16LE"
 #else
-#define INT_FORMAT "S16_BE"
+#define INT_FORMAT "S16BE"
 #endif
 
 GstStaticPadTemplate mulaw_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src",

diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c
index fbbdf36..f2d8193 100644 (file)
--- a/gst/spectrum/gstspectrum.c
+++ b/gst/spectrum/gstspectrum.c
@@ -111,9 +111,9 @@ GST_DEBUG_CATEGORY_STATIC (gst_spectrum_debug);
 
 /* elementfactory information */
 #if G_BYTE_ORDER == G_LITTLE_ENDIAN
-# define FORMATS "{ S16_LE, S24_3LE, S32_LE, F32_LE, F64_LE }"
+# define FORMATS "{ S16LE, S24LE, S32LE, F32LE, F64LE }"
 #else
-# define FORMATS "{ S16_BE, S24_3BE, S32_BE, F32_BE, F64_BE }"
+# define FORMATS "{ S16BE, S24BE, S32BE, F32BE, F64BE }"
 #endif
 
 #define ALLOWED_CAPS \
@@ -626,7 +626,7 @@ gst_spectrum_setup (GstAudioFilter * base, const GstAudioInfo * info)
       input_data =
           multi_channel ? input_data_int16_max : input_data_mixed_int16_max;
       break;
-    case GST_AUDIO_FORMAT_S24_3:
+    case GST_AUDIO_FORMAT_S24:
       input_data =
           multi_channel ? input_data_int24_max : input_data_mixed_int24_max;
       break;

diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c
index 3377eae..b04da8d 100644 (file)
--- a/gst/wavparse/gstwavparse.c
+++ b/gst/wavparse/gstwavparse.c
@@ -1743,7 +1743,7 @@ gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
 
   if (wav->caps) {
     s = gst_caps_get_structure (wav->caps, 0);
-    if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf != NULL) {
+    if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
       GstTypeFindProbability prob;
       GstCaps *tf_caps;