port more elements to new audio caps and API

diff --git a/ext/soup/gstsouphttpsrc.c b/ext/soup/gstsouphttpsrc.c
index 131fe74..e73ef51 100644 (file)
--- a/ext/soup/gstsouphttpsrc.c
+++ b/ext/soup/gstsouphttpsrc.c
@@ -802,13 +802,9 @@ gst_soup_http_src_got_headers_cb (SoupMessage * msg, GstSoupHTTPSrc * src)
       if (param != NULL)
         rate = atol (param);
 
       if (param != NULL)
         rate = atol (param);
 

-      src->src_caps = gst_caps_new_simple ("audio/x-raw-int",
-          "channels", G_TYPE_INT, channels,
-          "rate", G_TYPE_INT, rate,
-          "width", G_TYPE_INT, 16,
-          "depth", G_TYPE_INT, 16,
-          "signed", G_TYPE_BOOLEAN, TRUE,
-          "endianness", G_TYPE_INT, G_BIG_ENDIAN, NULL);
+      src->src_caps = gst_caps_new_simple ("audio/x-raw",
+          "format", G_TYPE_STRING, "S16_BE",
+          "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, rate, NULL);

     } else {
       /* Set the Content-Type field on the caps */
       if (src->src_caps)
     } else {
       /* Set the Content-Type field on the caps */
       if (src->src_caps)

diff --git a/ext/speex/gstspeexdec.c b/ext/speex/gstspeexdec.c
index 56a2afe..c5add82 100644 (file)
--- a/ext/speex/gstspeexdec.c
+++ b/ext/speex/gstspeexdec.c
@@ -46,6 +46,7 @@
 #include <stdlib.h>
 #include <string.h>
 #include <gst/tag/tag.h>
 #include <stdlib.h>
 #include <string.h>
 #include <gst/tag/tag.h>

+#include <gst/audio/audio.h>

 
 GST_DEBUG_CATEGORY_STATIC (speexdec_debug);
 #define GST_CAT_DEFAULT speexdec_debug
 
 GST_DEBUG_CATEGORY_STATIC (speexdec_debug);
 #define GST_CAT_DEFAULT speexdec_debug


@@ -58,15 +59,15 @@ enum
   ARG_ENH
 };
 
   ARG_ENH
 };
 

+#define FORMAT_STR GST_AUDIO_NE(S16)
+

 static GstStaticPadTemplate speex_dec_src_factory =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
 static GstStaticPadTemplate speex_dec_src_factory =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "rate = (int) [ 6000, 48000 ], "
-        "channels = (int) [ 1, 2 ], "
-        "endianness = (int) BYTE_ORDER, "
-        "signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " FORMAT_STR ", "
+        "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")

     );
 
 static GstStaticPadTemplate speex_dec_sink_factory =
     );
 
 static GstStaticPadTemplate speex_dec_sink_factory =


@@ -597,12 +598,10 @@ speex_dec_chain_parse_header (GstSpeexDec * dec, GstBuffer * buf)
   speex_bits_init (&dec->bits);
 
   /* set caps */
   speex_bits_init (&dec->bits);
 
   /* set caps */

-  caps = gst_caps_new_simple ("audio/x-raw-int",
+  caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, FORMAT_STR,

       "rate", G_TYPE_INT, dec->header->rate,
       "rate", G_TYPE_INT, dec->header->rate,

-      "channels", G_TYPE_INT, dec->header->nb_channels,
-      "signed", G_TYPE_BOOLEAN, TRUE,
-      "endianness", G_TYPE_INT, G_BYTE_ORDER,
-      "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
+      "channels", G_TYPE_INT, dec->header->nb_channels, NULL);

 
   if (!gst_pad_set_caps (dec->srcpad, caps))
     goto nego_failed;
 
   if (!gst_pad_set_caps (dec->srcpad, caps))
     goto nego_failed;

diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c
index 1da8cad..3b26866 100644 (file)
--- a/ext/speex/gstspeexenc.c
+++ b/ext/speex/gstspeexenc.c
@@ -52,14 +52,14 @@
 GST_DEBUG_CATEGORY_STATIC (speexenc_debug);
 #define GST_CAT_DEFAULT speexenc_debug
 
 GST_DEBUG_CATEGORY_STATIC (speexenc_debug);
 #define GST_CAT_DEFAULT speexenc_debug
 

+#define FORMAT_STR GST_AUDIO_NE(S16)
+

 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "rate = (int) [ 6000, 48000 ], "
-        "channels = (int) [ 1, 2 ], "
-        "endianness = (int) BYTE_ORDER, "
-        "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " FORMAT_STR ", "
+        "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")

     );
 
 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     );
 
 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",

diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c
index f39fcbf..054a640 100644 (file)
--- a/gst/auparse/gstauparse.c
+++ b/gst/auparse/gstauparse.c
@@ -43,6 +43,14 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_STATIC_CAPS ("audio/x-au")
     );
 
     GST_STATIC_CAPS ("audio/x-au")
     );
 

+#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 }, " \
+    "rate = (int) [ 8000, 192000 ], "       \
+    "channels = (int) [ 1, 2 ]"
+

 #define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \
     "audio/x-alaw, "                        \
     "rate = (int) [ 8000, 192000 ], "       \
 #define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \
     "audio/x-alaw, "                        \
     "rate = (int) [ 8000, 192000 ], "       \


@@ -61,8 +69,7 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
-        GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS ";"
+    GST_STATIC_CAPS (GST_AU_PARSE_RAW_PAD_TEMPLATE_CAPS "; "

         GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS ";"
         GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS ";"
         GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS));
         GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS ";"
         GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS ";"
         GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS));


@@ -180,7 +187,9 @@ gst_au_parse_parse_header (GstAuParse * auparse)
   guint32 size;
   guint8 *head;
   gchar layout[7] = { 0, };
   guint32 size;
   guint8 *head;
   gchar layout[7] = { 0, };

-  gint law = 0, depth = 0, ieee = 0;
+  GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
+  gint law = 0;
+  guint endianness;

 
   head = (guint8 *) gst_adapter_map (auparse->adapter, 24);
   g_assert (head != NULL);
 
   head = (guint8 *) gst_adapter_map (auparse->adapter, 24);
   g_assert (head != NULL);


@@ -190,14 +199,14 @@ gst_au_parse_parse_header (GstAuParse * auparse)
   switch (GST_READ_UINT32_BE (head)) {
       /* normal format is big endian (au is a Sparc format) */
     case 0x2e736e64:{          /* ".snd" */
   switch (GST_READ_UINT32_BE (head)) {
       /* normal format is big endian (au is a Sparc format) */
     case 0x2e736e64:{          /* ".snd" */

-      auparse->endianness = G_BIG_ENDIAN;
+      endianness = G_BIG_ENDIAN;

       break;
     }
       /* and of course, someone had to invent a little endian
        * version.  Used by DEC systems. */
     case 0x646e732e:           /* dns.                          */
     case 0x0064732e:{          /* other source say it is "dns." */
       break;
     }
       /* and of course, someone had to invent a little endian
        * version.  Used by DEC systems. */
     case 0x646e732e:           /* dns.                          */
     case 0x0064732e:{          /* other source say it is "dns." */

-      auparse->endianness = G_LITTLE_ENDIAN;
+      endianness = G_LITTLE_ENDIAN;

       break;
     }
     default:{
       break;
     }
     default:{


@@ -237,33 +246,50 @@ gst_au_parse_parse_header (GstAuParse * auparse)
   switch (auparse->encoding) {
     case 1:                    /* 8-bit ISDN mu-law G.711 */
       law = 1;
   switch (auparse->encoding) {
     case 1:                    /* 8-bit ISDN mu-law G.711 */
       law = 1;

-      depth = 8;

       break;
     case 27:                   /* 8-bit ISDN  A-law G.711 */
       law = 2;
       break;
     case 27:                   /* 8-bit ISDN  A-law G.711 */
       law = 2;

-      depth = 8;

       break;
 
       break;
 

-    case 2:                    /*  8-bit linear PCM */
-      depth = 8;
+    case 2:                    /*  8-bit linear PCM, FIXME signed? */
+      format = GST_AUDIO_FORMAT_S8;
+      auparse->sample_size = auparse->channels;

       break;
     case 3:                    /* 16-bit linear PCM */
       break;
     case 3:                    /* 16-bit linear PCM */

-      depth = 16;
+      if (endianness == G_LITTLE_ENDIAN)
+        format = GST_AUDIO_FORMAT_S16_LE;
+      else
+        format = GST_AUDIO_FORMAT_S16_BE;
+      auparse->sample_size = auparse->channels * 2;

       break;
     case 4:                    /* 24-bit linear PCM */
       break;
     case 4:                    /* 24-bit linear PCM */

-      depth = 24;
+      if (endianness == G_LITTLE_ENDIAN)
+        format = GST_AUDIO_FORMAT_S24_3LE;
+      else
+        format = GST_AUDIO_FORMAT_S24_3BE;
+      auparse->sample_size = auparse->channels * 3;

       break;
     case 5:                    /* 32-bit linear PCM */
       break;
     case 5:                    /* 32-bit linear PCM */

-      depth = 32;
+      if (endianness == G_LITTLE_ENDIAN)
+        format = GST_AUDIO_FORMAT_S32_LE;
+      else
+        format = GST_AUDIO_FORMAT_S32_BE;
+      auparse->sample_size = auparse->channels * 4;

       break;
 
     case 6:                    /* 32-bit IEEE floating point */
       break;
 
     case 6:                    /* 32-bit IEEE floating point */

-      ieee = 1;
-      depth = 32;
+      if (endianness == G_LITTLE_ENDIAN)
+        format = GST_AUDIO_FORMAT_F32_LE;
+      else
+        format = GST_AUDIO_FORMAT_F32_BE;
+      auparse->sample_size = auparse->channels * 4;

       break;
     case 7:                    /* 64-bit IEEE floating point */
       break;
     case 7:                    /* 64-bit IEEE floating point */

-      ieee = 1;
-      depth = 64;
+      if (endianness == G_LITTLE_ENDIAN)
+        format = GST_AUDIO_FORMAT_F64_LE;
+      else
+        format = GST_AUDIO_FORMAT_F64_BE;
+      auparse->sample_size = auparse->channels * 8;

       break;
 
     case 23:                   /* 4-bit CCITT G.721   ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */
       break;
 
     case 23:                   /* 4-bit CCITT G.721   ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */


@@ -308,27 +334,17 @@ gst_au_parse_parse_header (GstAuParse * auparse)
         "rate", G_TYPE_INT, auparse->samplerate,
         "channels", G_TYPE_INT, auparse->channels, NULL);
     auparse->sample_size = auparse->channels;
         "rate", G_TYPE_INT, auparse->samplerate,
         "channels", G_TYPE_INT, auparse->channels, NULL);
     auparse->sample_size = auparse->channels;

-  } else if (ieee) {
-    tempcaps = gst_caps_new_simple ("audio/x-raw-float",
+  } else if (format != GST_AUDIO_FORMAT_UNKNOWN) {
+    tempcaps = gst_caps_new_simple ("audio/x-raw",
+        "format", G_TYPE_STRING, gst_audio_format_to_string (format),

         "rate", G_TYPE_INT, auparse->samplerate,
         "rate", G_TYPE_INT, auparse->samplerate,

-        "channels", G_TYPE_INT, auparse->channels,
-        "endianness", G_TYPE_INT, auparse->endianness,
-        "width", G_TYPE_INT, depth, NULL);
-    auparse->sample_size = auparse->channels * depth / 8;
+        "channels", G_TYPE_INT, auparse->channels, NULL);

   } else if (layout[0]) {
     tempcaps = gst_caps_new_simple ("audio/x-adpcm",
         "layout", G_TYPE_STRING, layout, NULL);
     auparse->sample_size = 0;
   } else if (layout[0]) {
     tempcaps = gst_caps_new_simple ("audio/x-adpcm",
         "layout", G_TYPE_STRING, layout, NULL);
     auparse->sample_size = 0;

-  } else {
-    tempcaps = gst_caps_new_simple ("audio/x-raw-int",
-        "rate", G_TYPE_INT, auparse->samplerate,
-        "channels", G_TYPE_INT, auparse->channels,
-        "endianness", G_TYPE_INT, auparse->endianness,
-        "depth", G_TYPE_INT, depth, "width", G_TYPE_INT, depth,
-        /* FIXME: signed TRUE even for 8-bit PCM? */
-        "signed", G_TYPE_BOOLEAN, TRUE, NULL);
-    auparse->sample_size = auparse->channels * depth / 8;
-  }
+  } else
+    goto unknown_format;

 
   GST_DEBUG_OBJECT (auparse, "sample_size=%d", auparse->sample_size);
 
 
   GST_DEBUG_OBJECT (auparse, "sample_size=%d", auparse->sample_size);
 

diff --git a/gst/auparse/gstauparse.h b/gst/auparse/gstauparse.h
index 55d13e2..5744d26 100644 (file)
--- a/gst/auparse/gstauparse.h
+++ b/gst/auparse/gstauparse.h
@@ -60,7 +60,6 @@ struct _GstAuParse {
   guint       sample_size;
   guint       encoding;
   guint       samplerate;
   guint       sample_size;
   guint       encoding;
   guint       samplerate;

-  guint       endianness;

   guint       channels;
 };
 
   guint       channels;
 };
 

diff --git a/gst/cutter/gstcutter.c b/gst/cutter/gstcutter.c
index 98073c5..32816da 100644 (file)
--- a/gst/cutter/gstcutter.c
+++ b/gst/cutter/gstcutter.c
@@ -68,12 +68,9 @@ GST_DEBUG_CATEGORY_STATIC (cutter_debug);
 static GstStaticPadTemplate cutter_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
 static GstStaticPadTemplate cutter_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "endianness = (int) BYTE_ORDER, "
-        "width = (int) { 8, 16 }, "
-        "depth = (int) { 8, 16 }, " "signed = (boolean) true")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { " GST_AUDIO_NE (S8) "," GST_AUDIO_NE (S16) " }, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")

     );
 
 static GstStaticPadTemplate cutter_sink_factory =
     );
 
 static GstStaticPadTemplate cutter_sink_factory =


@@ -81,11 +78,8 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "

-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "endianness = (int) BYTE_ORDER, "
-        "width = (int) { 8, 16 }, "
-        "depth = (int) { 8, 16 }, " "signed = (boolean) true")
+        "format = (string) { " GST_AUDIO_NE (S8) "," GST_AUDIO_NE (S16) " }, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")

     );
 
 enum
     );
 
 enum


@@ -363,6 +357,7 @@ gst_cutter_get_caps (GstPad * pad, GstCutter * filter)
 {
   GstCaps *caps;
   GstStructure *structure;
 {
   GstCaps *caps;
   GstStructure *structure;

+  const gchar *format;

 
   caps = gst_pad_get_current_caps (pad);
   if (!caps) {
 
   caps = gst_pad_get_current_caps (pad);
   if (!caps) {


@@ -370,8 +365,12 @@ gst_cutter_get_caps (GstPad * pad, GstCutter * filter)
     return FALSE;
   }
   structure = gst_caps_get_structure (caps, 0);
     return FALSE;
   }
   structure = gst_caps_get_structure (caps, 0);

-  gst_structure_get_int (structure, "width", &filter->width);
-  filter->max_sample = 1 << (filter->width - 1);        /* signed */
+
+  format = gst_structure_get_string (structure, "format");
+  if (g_str_has_prefix (format, "S16"))
+    filter->width = 16;
+  else
+    filter->width = 8;

   filter->have_caps = TRUE;
 
   gst_caps_unref (caps);
   filter->have_caps = TRUE;
 
   gst_caps_unref (caps);

diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c
index 86b168d..e516c99 100644 (file)
--- a/gst/equalizer/gstiirequalizer.c
+++ b/gst/equalizer/gstiirequalizer.c
@@ -43,22 +43,15 @@ 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,
 static void gst_iir_equalizer_finalize (GObject * object);
 
 static gboolean gst_iir_equalizer_setup (GstAudioFilter * filter,

-    GstRingBufferSpec * fmt);
+    GstAudioInfo * info);

 static GstFlowReturn gst_iir_equalizer_transform_ip (GstBaseTransform * btrans,
     GstBuffer * buf);
 
 #define ALLOWED_CAPS \
 static GstFlowReturn gst_iir_equalizer_transform_ip (GstBaseTransform * btrans,
     GstBuffer * buf);
 
 #define ALLOWED_CAPS \

-    "audio/x-raw-int,"                                                \
-    " depth=(int)16,"                                                 \
-    " width=(int)16,"                                                 \
-    " endianness=(int)BYTE_ORDER,"                                    \
-    " signed=(bool)TRUE,"                                             \
-    " rate=(int)[1000,MAX],"                                          \
-    " channels=(int)[1,MAX]; "                                        \
-    "audio/x-raw-float,"                                              \
-    " width=(int) { 32, 64 } ,"                                       \
-    " endianness=(int)BYTE_ORDER,"                                    \
-    " rate=(int)[1000,MAX],"                                          \
+    "audio/x-raw,"                                                \
+    " format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)","  \
+                        GST_AUDIO_NE(F64)" }, "                   \
+    " rate=(int)[1000,MAX],"                                      \

     " channels=(int)[1,MAX]"
 
 #define gst_iir_equalizer_parent_class parent_class
     " channels=(int)[1,MAX]"
 
 #define gst_iir_equalizer_parent_class parent_class


@@ -465,15 +458,17 @@ calculate_bw (GstIirEqualizerBand * band, gint rate)
 static void
 setup_peak_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 {
 static void
 setup_peak_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 {

-  g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
+  gint rate = GST_AUDIO_FILTER_RATE (equ);
+
+  g_return_if_fail (rate);

 
   {
     gdouble gain, omega, bw;
     gdouble alpha, alpha1, alpha2, b0;
 
     gain = arg_to_scale (band->gain);
 
   {
     gdouble gain, omega, bw;
     gdouble alpha, alpha1, alpha2, b0;
 
     gain = arg_to_scale (band->gain);

-    omega = calculate_omega (band->freq, GST_AUDIO_FILTER (equ)->format.rate);
-    bw = calculate_bw (band, GST_AUDIO_FILTER (equ)->format.rate);
+    omega = calculate_omega (band->freq, rate);
+    bw = calculate_bw (band, rate);

     if (bw == 0.0)
       goto out;
 
     if (bw == 0.0)
       goto out;
 


@@ -501,7 +496,9 @@ setup_peak_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 static void
 setup_low_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 {
 static void
 setup_low_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 {

-  g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
+  gint rate = GST_AUDIO_FILTER_RATE (equ);
+
+  g_return_if_fail (rate);

 
   {
     gdouble gain, omega, bw;
 
   {
     gdouble gain, omega, bw;


@@ -509,8 +506,8 @@ setup_low_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
     gdouble egp, egm;
 
     gain = arg_to_scale (band->gain);
     gdouble egp, egm;
 
     gain = arg_to_scale (band->gain);

-    omega = calculate_omega (band->freq, GST_AUDIO_FILTER (equ)->format.rate);
-    bw = calculate_bw (band, GST_AUDIO_FILTER (equ)->format.rate);
+    omega = calculate_omega (band->freq, rate);
+    bw = calculate_bw (band, rate);

     if (bw == 0.0)
       goto out;
 
     if (bw == 0.0)
       goto out;
 


@@ -539,7 +536,9 @@ setup_low_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 static void
 setup_high_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 {
 static void
 setup_high_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
 {

-  g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
+  gint rate = GST_AUDIO_FILTER_RATE (equ);
+
+  g_return_if_fail (rate);

 
   {
     gdouble gain, omega, bw;
 
   {
     gdouble gain, omega, bw;


@@ -547,8 +546,8 @@ setup_high_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
     gdouble egp, egm;
 
     gain = arg_to_scale (band->gain);
     gdouble egp, egm;
 
     gain = arg_to_scale (band->gain);

-    omega = calculate_omega (band->freq, GST_AUDIO_FILTER (equ)->format.rate);
-    bw = calculate_bw (band, GST_AUDIO_FILTER (equ)->format.rate);
+    omega = calculate_omega (band->freq, rate);
+    bw = calculate_bw (band, rate);

     if (bw == 0.0)
       goto out;
 
     if (bw == 0.0)
       goto out;
 


@@ -614,7 +613,7 @@ alloc_history (GstIirEqualizer * equ)
   /* free + alloc = no memcpy */
   g_free (equ->history);
   equ->history =
   /* free + alloc = no memcpy */
   g_free (equ->history);
   equ->history =

-      g_malloc0 (equ->history_size * GST_AUDIO_FILTER (equ)->format.channels *
+      g_malloc0 (equ->history_size * GST_AUDIO_FILTER_CHANNELS (equ) *

       equ->freq_band_count);
 }
 
       equ->freq_band_count);
 }
 


@@ -819,8 +818,9 @@ gst_iir_equalizer_transform_ip (GstBaseTransform * btrans, GstBuffer * buf)
   GstClockTime timestamp;
   guint8 *data;
   gsize size;
   GstClockTime timestamp;
   guint8 *data;
   gsize size;

+  gint channels = GST_AUDIO_FILTER_CHANNELS (filter);

 
 

-  if (G_UNLIKELY (filter->format.channels < 1 || equ->process == NULL))
+  if (G_UNLIKELY (channels < 1 || equ->process == NULL))

     return GST_FLOW_NOT_NEGOTIATED;
 
   BANDS_LOCK (equ);
     return GST_FLOW_NOT_NEGOTIATED;
 
   BANDS_LOCK (equ);


@@ -841,41 +841,29 @@ gst_iir_equalizer_transform_ip (GstBaseTransform * btrans, GstBuffer * buf)
     gst_object_sync_values (G_OBJECT (equ), timestamp);
 
   data = gst_buffer_map (buf, &size, NULL, GST_MAP_WRITE);
     gst_object_sync_values (G_OBJECT (equ), timestamp);
 
   data = gst_buffer_map (buf, &size, NULL, GST_MAP_WRITE);

-  equ->process (equ, data, size, filter->format.channels);
+  equ->process (equ, data, size, channels);

   gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }
 
 static gboolean
   gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }
 
 static gboolean

-gst_iir_equalizer_setup (GstAudioFilter * audio, GstRingBufferSpec * fmt)
+gst_iir_equalizer_setup (GstAudioFilter * audio, GstAudioInfo * info)

 {
   GstIirEqualizer *equ = GST_IIR_EQUALIZER (audio);
 
 {
   GstIirEqualizer *equ = GST_IIR_EQUALIZER (audio);
 

-  switch (fmt->type) {
-    case GST_BUFTYPE_LINEAR:
-      switch (fmt->width) {
-        case 16:
-          equ->history_size = history_size_gint16;
-          equ->process = gst_iir_equ_process_gint16;
-          break;
-        default:
-          return FALSE;
-      }
+  switch (GST_AUDIO_INFO_FORMAT (info)) {
+    case GST_AUDIO_FORMAT_S16:
+      equ->history_size = history_size_gint16;
+      equ->process = gst_iir_equ_process_gint16;

       break;
       break;

-    case GST_BUFTYPE_FLOAT:
-      switch (fmt->width) {
-        case 32:
-          equ->history_size = history_size_gfloat;
-          equ->process = gst_iir_equ_process_gfloat;
-          break;
-        case 64:
-          equ->history_size = history_size_gdouble;
-          equ->process = gst_iir_equ_process_gdouble;
-          break;
-        default:
-          return FALSE;
-      }
+    case GST_AUDIO_FORMAT_F32:
+      equ->history_size = history_size_gfloat;
+      equ->process = gst_iir_equ_process_gfloat;
+      break;
+    case GST_AUDIO_FORMAT_F64:
+      equ->history_size = history_size_gdouble;
+      equ->process = gst_iir_equ_process_gdouble;

       break;
     default:
       return FALSE;
       break;
     default:
       return FALSE;

diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c
index 28977bc..16efded 100644 (file)
--- a/gst/level/gstlevel.c
+++ b/gst/level/gstlevel.c
@@ -117,37 +117,23 @@ GST_DEBUG_CATEGORY_STATIC (level_debug);
 #define EPSILON 1e-35f
 
 static GstStaticPadTemplate sink_template_factory =
 #define EPSILON 1e-35f
 
 static GstStaticPadTemplate sink_template_factory =

-    GST_STATIC_PAD_TEMPLATE ("sink",
+GST_STATIC_PAD_TEMPLATE ("sink",

     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_PAD_SINK,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "endianness = (int) BYTE_ORDER, "
-        "width = (int) { 8, 16, 32 }, "
-        "depth = (int) { 8, 16, 32 }, "
-        "signed = (boolean) true; "
-        "audio/x-raw-float, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "endianness = (int) BYTE_ORDER, " "width = (int) {32, 64} ")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
+        GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")

     );
 
 static GstStaticPadTemplate src_template_factory =
     );
 
 static GstStaticPadTemplate src_template_factory =

-    GST_STATIC_PAD_TEMPLATE ("src",
+GST_STATIC_PAD_TEMPLATE ("src",

     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "endianness = (int) BYTE_ORDER, "
-        "width = (int) { 8, 16, 32 }, "
-        "depth = (int) { 8, 16, 32 }, "
-        "signed = (boolean) true; "
-        "audio/x-raw-float, "
-        "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, MAX ], "
-        "endianness = (int) BYTE_ORDER, " "width = (int) {32, 64} ")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { S8, " GST_AUDIO_NE (S16) ", " GST_AUDIO_NE (S32)
+        GST_AUDIO_NE (F32) "," GST_AUDIO_NE (F64) " },"
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")

     );
 
 enum
     );
 
 enum


@@ -228,9 +214,7 @@ gst_level_init (GstLevel * filter)
   filter->CS = NULL;
   filter->peak = NULL;
 
   filter->CS = NULL;
   filter->peak = NULL;
 

-  filter->rate = 0;
-  filter->width = 0;
-  filter->channels = 0;
+  gst_audio_info_init (&filter->info);

 
   filter->interval = GST_SECOND / 10;
   filter->decay_peak_ttl = GST_SECOND / 10 * 3;
 
   filter->interval = GST_SECOND / 10;
   filter->decay_peak_ttl = GST_SECOND / 10 * 3;


@@ -277,9 +261,10 @@ gst_level_set_property (GObject * object, guint prop_id,
       break;
     case PROP_SIGNAL_INTERVAL:
       filter->interval = g_value_get_uint64 (value);
       break;
     case PROP_SIGNAL_INTERVAL:
       filter->interval = g_value_get_uint64 (value);

-      if (filter->rate) {
+      if (GST_AUDIO_INFO_RATE (&filter->info)) {

         filter->interval_frames =
         filter->interval_frames =

-            GST_CLOCK_TIME_TO_FRAMES (filter->interval, filter->rate);
+            GST_CLOCK_TIME_TO_FRAMES (filter->interval,
+            GST_AUDIO_INFO_RATE (&filter->info));

       }
       break;
     case PROP_PEAK_TTL:
       }
       break;
     case PROP_PEAK_TTL:


@@ -410,58 +395,42 @@ gst_level_calculate_gdouble (gpointer data, guint num, guint channels,
 */
 
 
 */
 
 

-static gint
-structure_get_int (GstStructure * structure, const gchar * field)
-{
-  gint ret;
-
-  if (!gst_structure_get_int (structure, field, &ret))
-    g_assert_not_reached ();
-
-  return ret;
-}
-

 static gboolean
 gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
 {
   GstLevel *filter = GST_LEVEL (trans);
 static gboolean
 gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
 {
   GstLevel *filter = GST_LEVEL (trans);

-  const gchar *mimetype;
-  GstStructure *structure;
-  gint i;
+  GstAudioInfo info;
+  gint i, channels, rate;

 
 

-  structure = gst_caps_get_structure (in, 0);
-  filter->rate = structure_get_int (structure, "rate");
-  filter->width = structure_get_int (structure, "width");
-  filter->channels = structure_get_int (structure, "channels");
-  mimetype = gst_structure_get_name (structure);
+  if (!gst_audio_info_from_caps (&info, in))
+    return FALSE;

 
 

-  /* FIXME: set calculator func depending on caps */
-  filter->process = NULL;
-  if (strcmp (mimetype, "audio/x-raw-int") == 0) {
-    GST_DEBUG_OBJECT (filter, "use int: %u", filter->width);
-    switch (filter->width) {
-      case 8:
-        filter->process = gst_level_calculate_gint8;
-        break;
-      case 16:
-        filter->process = gst_level_calculate_gint16;
-        break;
-      case 32:
-        filter->process = gst_level_calculate_gint32;
-        break;
-    }
-  } else if (strcmp (mimetype, "audio/x-raw-float") == 0) {
-    GST_DEBUG_OBJECT (filter, "use float, %u", filter->width);
-    switch (filter->width) {
-      case 32:
-        filter->process = gst_level_calculate_gfloat;
-        break;
-      case 64:
-        filter->process = gst_level_calculate_gdouble;
-        break;
-    }
+  switch (GST_AUDIO_INFO_FORMAT (&info)) {
+    case GST_AUDIO_FORMAT_S8:
+      filter->process = gst_level_calculate_gint8;
+      break;
+    case GST_AUDIO_FORMAT_S16:
+      filter->process = gst_level_calculate_gint16;
+      break;
+    case GST_AUDIO_FORMAT_S32:
+      filter->process = gst_level_calculate_gint32;
+      break;
+    case GST_AUDIO_FORMAT_F32:
+      filter->process = gst_level_calculate_gfloat;
+      break;
+    case GST_AUDIO_FORMAT_F64:
+      filter->process = gst_level_calculate_gdouble;
+      break;
+    default:
+      filter->process = NULL;
+      break;

   }
 
   }
 

+  filter->info = info;
+
+  channels = GST_AUDIO_INFO_CHANNELS (&info);
+  rate = GST_AUDIO_INFO_RATE (&info);
+

   /* allocate channel variable arrays */
   g_free (filter->CS);
   g_free (filter->peak);
   /* allocate channel variable arrays */
   g_free (filter->CS);
   g_free (filter->peak);


@@ -469,22 +438,21 @@ gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
   g_free (filter->decay_peak);
   g_free (filter->decay_peak_base);
   g_free (filter->decay_peak_age);
   g_free (filter->decay_peak);
   g_free (filter->decay_peak_base);
   g_free (filter->decay_peak_age);

-  filter->CS = g_new (gdouble, filter->channels);
-  filter->peak = g_new (gdouble, filter->channels);
-  filter->last_peak = g_new (gdouble, filter->channels);
-  filter->decay_peak = g_new (gdouble, filter->channels);
-  filter->decay_peak_base = g_new (gdouble, filter->channels);
+  filter->CS = g_new (gdouble, channels);
+  filter->peak = g_new (gdouble, channels);
+  filter->last_peak = g_new (gdouble, channels);
+  filter->decay_peak = g_new (gdouble, channels);
+  filter->decay_peak_base = g_new (gdouble, channels);

 
 

-  filter->decay_peak_age = g_new (GstClockTime, filter->channels);
+  filter->decay_peak_age = g_new (GstClockTime, channels);

 
 

-  for (i = 0; i < filter->channels; ++i) {
+  for (i = 0; i < channels; ++i) {

     filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
         filter->decay_peak[i] = filter->decay_peak_base[i] = 0.0;
     filter->decay_peak_age[i] = G_GUINT64_CONSTANT (0);
   }
 
     filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
         filter->decay_peak[i] = filter->decay_peak_base[i] = 0.0;
     filter->decay_peak_age[i] = G_GUINT64_CONSTANT (0);
   }
 

-  filter->interval_frames =
-      GST_CLOCK_TIME_TO_FRAMES (filter->interval, filter->rate);
+  filter->interval_frames = GST_CLOCK_TIME_TO_FRAMES (filter->interval, rate);

 
   return TRUE;
 }
 
   return TRUE;
 }


@@ -572,35 +540,38 @@ gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
   guint num_int_samples = 0;    /* number of interleaved samples
                                  * ie. total count for all channels combined */
   GstClockTimeDiff falloff_time;
   guint num_int_samples = 0;    /* number of interleaved samples
                                  * ie. total count for all channels combined */
   GstClockTimeDiff falloff_time;

+  gint channels, rate, bps;

 
   filter = GST_LEVEL (trans);
 
 
   filter = GST_LEVEL (trans);
 

+  channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
+  bps = GST_AUDIO_INFO_BPS (&filter->info);
+  rate = GST_AUDIO_INFO_RATE (&filter->info);
+

   in_data = data = gst_buffer_map (in, &in_size, NULL, GST_MAP_READ);
   in_data = data = gst_buffer_map (in, &in_size, NULL, GST_MAP_READ);

-  num_int_samples = in_size / (filter->width / 8);
+  num_int_samples = in_size / bps;

 
   GST_LOG_OBJECT (filter, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
       num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (in)));
 
 
   GST_LOG_OBJECT (filter, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
       num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (in)));
 

-  g_return_val_if_fail (num_int_samples % filter->channels == 0,
-      GST_FLOW_ERROR);
+  g_return_val_if_fail (num_int_samples % channels == 0, GST_FLOW_ERROR);

 
 

-  num_frames = num_int_samples / filter->channels;
+  num_frames = num_int_samples / channels;

 
 

-  for (i = 0; i < filter->channels; ++i) {
+  for (i = 0; i < channels; ++i) {

     if (!GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_GAP)) {
     if (!GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_GAP)) {

-      filter->process (in_data, num_int_samples, filter->channels, &CS,
+      filter->process (in_data, num_int_samples, channels, &CS,

           &filter->peak[i]);
       GST_LOG_OBJECT (filter,
           "channel %d, cumulative sum %f, peak %f, over %d samples/%d channels",
           &filter->peak[i]);
       GST_LOG_OBJECT (filter,
           "channel %d, cumulative sum %f, peak %f, over %d samples/%d channels",

-          i, CS, filter->peak[i], num_int_samples, filter->channels);
+          i, CS, filter->peak[i], num_int_samples, channels);

       filter->CS[i] += CS;
     } else {
       filter->peak[i] = 0.0;
     }
       filter->CS[i] += CS;
     } else {
       filter->peak[i] = 0.0;
     }

-    in_data += (filter->width / 8);
+    in_data += bps;

 
 

-    filter->decay_peak_age[i] +=
-        GST_FRAMES_TO_CLOCK_TIME (num_frames, filter->rate);
+    filter->decay_peak_age[i] += GST_FRAMES_TO_CLOCK_TIME (num_frames, rate);

     GST_LOG_OBJECT (filter, "filter peak info [%d]: decay peak %f, age %"
         GST_TIME_FORMAT, i,
         filter->decay_peak[i], GST_TIME_ARGS (filter->decay_peak_age[i]));
     GST_LOG_OBJECT (filter, "filter peak info [%d]: decay peak %f, age %"
         GST_TIME_FORMAT, i,
         filter->decay_peak[i], GST_TIME_ARGS (filter->decay_peak_age[i]));


@@ -656,7 +627,7 @@ gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
     if (filter->message) {
       GstMessage *m;
       GstClockTime duration =
     if (filter->message) {
       GstMessage *m;
       GstClockTime duration =

-          GST_FRAMES_TO_CLOCK_TIME (filter->num_frames, filter->rate);
+          GST_FRAMES_TO_CLOCK_TIME (filter->num_frames, rate);

 
       m = gst_level_message_new (filter, filter->message_ts, duration);
 
 
       m = gst_level_message_new (filter, filter->message_ts, duration);
 


@@ -664,7 +635,7 @@ gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
           "message: ts %" GST_TIME_FORMAT ", num_frames %d",
           GST_TIME_ARGS (filter->message_ts), filter->num_frames);
 
           "message: ts %" GST_TIME_FORMAT ", num_frames %d",
           GST_TIME_ARGS (filter->message_ts), filter->num_frames);
 

-      for (i = 0; i < filter->channels; ++i) {
+      for (i = 0; i < channels; ++i) {

         gdouble RMS;
         gdouble RMSdB, lastdB, decaydB;
 
         gdouble RMS;
         gdouble RMSdB, lastdB, decaydB;
 

diff --git a/gst/level/gstlevel.h b/gst/level/gstlevel.h
index 32d4c9e..ae9ba3a 100644 (file)
--- a/gst/level/gstlevel.h
+++ b/gst/level/gstlevel.h
@@ -26,7 +26,7 @@
 
 #include <gst/gst.h>
 #include <gst/base/gstbasetransform.h>
 
 #include <gst/gst.h>
 #include <gst/base/gstbasetransform.h>

-
+#include <gst/audio/audio.h>

 
 G_BEGIN_DECLS
 
 
 G_BEGIN_DECLS
 


@@ -59,9 +59,7 @@ struct _GstLevel {
   gboolean message;             /* whether or not to post messages */
   guint64 interval;             /* how many seconds between emits */
 
   gboolean message;             /* whether or not to post messages */
   guint64 interval;             /* how many seconds between emits */
 

-  gint rate;                    /* caps variables */
-  gint width;
-  gint channels;
+  GstAudioInfo info;

 
   gdouble decay_peak_ttl;       /* time to live for peak in seconds */
   gdouble decay_peak_falloff;   /* falloff in dB/sec */
 
   gdouble decay_peak_ttl;       /* time to live for peak in seconds */
   gdouble decay_peak_falloff;   /* falloff in dB/sec */


@@ -79,7 +77,7 @@ struct _GstLevel {
   gdouble *MS;                  /* normalized Mean Square of buffer */
   gdouble *RMS_dB;              /* RMS in dB to emit */
   GstClockTime *decay_peak_age; /* age of last peak */
   gdouble *MS;                  /* normalized Mean Square of buffer */
   gdouble *RMS_dB;              /* RMS in dB to emit */
   GstClockTime *decay_peak_age; /* age of last peak */

-  
+

   void (*process)(gpointer, guint, guint, gdouble*, gdouble*);
 };
 
   void (*process)(gpointer, guint, guint, gdouble*, gdouble*);
 };
 

diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c
index 5534544..1321a7e 100644 (file)
--- a/gst/rtp/gstrtpL16depay.c
+++ b/gst/rtp/gstrtpL16depay.c
@@ -37,11 +37,8 @@ static GstStaticPadTemplate gst_rtp_L16_depay_src_template =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) BIG_ENDIAN, "
-        "signed = (boolean) true, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) S16_BE, "

         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
 
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
 


@@ -177,11 +174,8 @@ gst_rtp_L16_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
   rtpL16depay->rate = clock_rate;
   rtpL16depay->channels = channels;
 
   rtpL16depay->rate = clock_rate;
   rtpL16depay->channels = channels;
 

-  srccaps = gst_caps_new_simple ("audio/x-raw-int",
-      "endianness", G_TYPE_INT, G_BIG_ENDIAN,
-      "signed", G_TYPE_BOOLEAN, TRUE,
-      "width", G_TYPE_INT, 16,
-      "depth", G_TYPE_INT, 16,
+  srccaps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, "S16_BE",

       "rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL);
 
   /* add channel positions */
       "rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL);
 
   /* add channel positions */

diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index 2d4543b..e13e135 100644 (file)
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -37,11 +37,8 @@ static GstStaticPadTemplate gst_rtp_L16_pay_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) BIG_ENDIAN, "
-        "signed = (boolean) true, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) S16_BE, "

         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
 
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
 

diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c
index eb27bcf..e32f450 100644 (file)
--- a/gst/rtp/gstrtpvrawdepay.c
+++ b/gst/rtp/gstrtpvrawdepay.c
@@ -31,10 +31,10 @@ GST_DEBUG_CATEGORY_STATIC (rtpvrawdepay_debug);
 #define GST_CAT_DEFAULT (rtpvrawdepay_debug)
 
 static GstStaticPadTemplate gst_rtp_vraw_depay_src_template =
 #define GST_CAT_DEFAULT (rtpvrawdepay_debug)
 
 static GstStaticPadTemplate gst_rtp_vraw_depay_src_template =

-    GST_STATIC_PAD_TEMPLATE ("src",
+GST_STATIC_PAD_TEMPLATE ("src",

     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("video/x-raw-rgb; video/x-raw-yuv")
+    GST_STATIC_CAPS ("video/x-raw")

     );
 
 static GstStaticPadTemplate gst_rtp_vraw_depay_sink_template =
     );
 
 static GstStaticPadTemplate gst_rtp_vraw_depay_sink_template =

diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c
index f127c36..a74d1f0 100644 (file)
--- a/gst/spectrum/gstspectrum.c
+++ b/gst/spectrum/gstspectrum.c
@@ -110,37 +110,17 @@ GST_DEBUG_CATEGORY_STATIC (gst_spectrum_debug);
 #define GST_CAT_DEFAULT gst_spectrum_debug
 
 /* elementfactory information */
 #define GST_CAT_DEFAULT gst_spectrum_debug
 
 /* elementfactory information */

+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+# define FORMATS "{ S16_LE, S24_3LE, S32_LE, F32_LE, F64_LE }"
+#else
+# define FORMATS "{ S16_BE, S24_3BE, S32_BE, F32_BE, F64_BE }"
+#endif

 
 #define ALLOWED_CAPS \
 
 #define ALLOWED_CAPS \

-    "audio/x-raw-int, "                                               \
-    " width = (int) 16, "                                             \
-    " depth = (int) [ 1, 16 ], "                                      \
-    " signed = (boolean) true, "                                      \
-    " endianness = (int) BYTE_ORDER, "                                \
-    " rate = (int) [ 1, MAX ], "                                      \
-    " channels = (int) [ 1, MAX ]; "                                  \
-    "audio/x-raw-int, "                                               \
-    " width = (int) 24, "                                             \
-    " depth = (int) [ 1, 24 ], "                                      \
-    " signed = (boolean) true, "                                      \
-    " endianness = (int) BYTE_ORDER, "                                \
-    " rate = (int) [ 1, MAX ], "                                      \
-    " channels = (int) [ 1, MAX ]; "                                  \
-    "audio/x-raw-int, "                                               \
-    " width = (int) 32, "                                             \
-    " depth = (int) [ 1, 32 ], "                                      \
-    " signed = (boolean) true, "                                      \
-    " endianness = (int) BYTE_ORDER, "                                \
-    " rate = (int) [ 1, MAX ], "                                      \
-    " channels = (int) [ 1, MAX ]; "                                  \
-    "audio/x-raw-float, "                                             \
-    " width = (int) { 32, 64 }, "                                     \
-    " endianness = (int) BYTE_ORDER, "                                \
-    " rate = (int) [ 1, MAX ], "                                      \
-    " channels = (int) [ 1, MAX ]"
+  GST_AUDIO_CAPS_MAKE (FORMATS)

 
 /* Spectrum properties */
 
 /* Spectrum properties */

-#define DEFAULT_POST_MESSAGES                  TRUE
+#define DEFAULT_POST_MESSAGES          TRUE

 #define DEFAULT_MESSAGE_MAGNITUDE      TRUE
 #define DEFAULT_MESSAGE_PHASE          FALSE
 #define DEFAULT_INTERVAL               (GST_SECOND / 10)
 #define DEFAULT_MESSAGE_MAGNITUDE      TRUE
 #define DEFAULT_MESSAGE_PHASE          FALSE
 #define DEFAULT_INTERVAL               (GST_SECOND / 10)


@@ -172,8 +152,7 @@ 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_stop (GstBaseTransform * trans);
 static GstFlowReturn gst_spectrum_transform_ip (GstBaseTransform * trans,
     GstBuffer * in);

-static gboolean gst_spectrum_setup (GstAudioFilter * base,
-    GstRingBufferSpec * format);
+static gboolean gst_spectrum_setup (GstAudioFilter * base, GstAudioInfo * info);

 
 static void
 gst_spectrum_class_init (GstSpectrumClass * klass)
 
 static void
 gst_spectrum_class_init (GstSpectrumClass * klass)


@@ -287,7 +266,7 @@ gst_spectrum_alloc_channel_data (GstSpectrum * spectrum)
   g_assert (spectrum->channel_data == NULL);
 
   spectrum->num_channels = (spectrum->multi_channel) ?
   g_assert (spectrum->channel_data == NULL);
 
   spectrum->num_channels = (spectrum->multi_channel) ?

-      GST_AUDIO_FILTER (spectrum)->format.channels : 1;
+      GST_AUDIO_FILTER_CHANNELS (spectrum) : 1;

 
   GST_DEBUG_OBJECT (spectrum, "allocating data for %d channels",
       spectrum->num_channels);
 
   GST_DEBUG_OBJECT (spectrum, "allocating data for %d channels",
       spectrum->num_channels);


@@ -503,23 +482,6 @@ input_data_mixed_double (const guint8 * _in, gfloat * out, guint len,
 }
 
 static void
 }
 
 static void

-input_data_mixed_int32 (const guint8 * _in, gfloat * out, guint len,
-    guint channels, gfloat max_value, guint op, guint nfft)
-{
-  guint i, j, ip = 0;
-  gint32 *in = (gint32 *) _in;
-  gfloat v;
-
-  for (j = 0; j < len; j++) {
-    v = in[ip++] * 2 + 1;
-    for (i = 1; i < channels; i++)
-      v += in[ip++] * 2 + 1;
-    out[op] = v / channels;
-    op = (op + 1) % nfft;
-  }
-}
-
-static void

 input_data_mixed_int32_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {
 input_data_mixed_int32_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {


@@ -537,30 +499,6 @@ input_data_mixed_int32_max (const guint8 * _in, gfloat * out, guint len,
 }
 
 static void
 }
 
 static void

-input_data_mixed_int24 (const guint8 * _in, gfloat * out, guint len,
-    guint channels, gfloat max_value, guint op, guint nfft)
-{
-  guint i, j;
-  gfloat v = 0.0;
-
-  for (j = 0; j < len; j++) {
-    for (i = 0; i < channels; i++) {
-#if G_BYTE_ORDER == G_BIG_ENDIAN
-      gint32 value = GST_READ_UINT24_BE (_in);
-#else
-      gint32 value = GST_READ_UINT24_LE (_in);
-#endif
-      if (value & 0x00800000)
-        value |= 0xff000000;
-      v += value * 2 + 1;
-      _in += 3;
-    }
-    out[op] = v / channels;
-    op = (op + 1) % nfft;
-  }
-}
-
-static void

 input_data_mixed_int24_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {
 input_data_mixed_int24_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {


@@ -585,23 +523,6 @@ input_data_mixed_int24_max (const guint8 * _in, gfloat * out, guint len,
 }
 
 static void
 }
 
 static void

-input_data_mixed_int16 (const guint8 * _in, gfloat * out, guint len,
-    guint channels, gfloat max_value, guint op, guint nfft)
-{
-  guint i, j, ip = 0;
-  gint16 *in = (gint16 *) _in;
-  gfloat v;
-
-  for (j = 0; j < len; j++) {
-    v = in[ip++] * 2 + 1;
-    for (i = 1; i < channels; i++)
-      v += in[ip++] * 2 + 1;
-    out[op] = v / channels;
-    op = (op + 1) % nfft;
-  }
-}
-
-static void

 input_data_mixed_int16_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {
 input_data_mixed_int16_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {


@@ -647,19 +568,6 @@ input_data_double (const guint8 * _in, gfloat * out, guint len, guint channels,
 }
 
 static void
 }
 
 static void

-input_data_int32 (const guint8 * _in, gfloat * out, guint len, guint channels,
-    gfloat max_value, guint op, guint nfft)
-{
-  guint j, ip;
-  gint32 *in = (gint32 *) _in;
-
-  for (j = 0, ip = 0; j < len; j++, ip += channels) {
-    out[op] = in[ip] * 2 + 1;
-    op = (op + 1) % nfft;
-  }
-}
-
-static void

 input_data_int32_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {
 input_data_int32_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {


@@ -673,26 +581,6 @@ input_data_int32_max (const guint8 * _in, gfloat * out, guint len,
 }
 
 static void
 }
 
 static void

-input_data_int24 (const guint8 * _in, gfloat * out, guint len, guint channels,
-    gfloat max_value, guint op, guint nfft)
-{
-  guint j;
-
-  for (j = 0; j < len; j++) {
-#if G_BYTE_ORDER == G_BIG_ENDIAN
-    gint32 v = GST_READ_UINT24_BE (_in);
-#else
-    gint32 v = GST_READ_UINT24_LE (_in);
-#endif
-    if (v & 0x00800000)
-      v |= 0xff000000;
-    _in += 3 * channels;
-    out[op] = v * 2 + 1;
-    op = (op + 1) % nfft;
-  }
-}
-
-static void

 input_data_int24_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {
 input_data_int24_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {


@@ -713,19 +601,6 @@ input_data_int24_max (const guint8 * _in, gfloat * out, guint len,
 }
 
 static void
 }
 
 static void

-input_data_int16 (const guint8 * _in, gfloat * out, guint len, guint channels,
-    gfloat max_value, guint op, guint nfft)
-{
-  guint j, ip;
-  gint16 *in = (gint16 *) _in;
-
-  for (j = 0, ip = 0; j < len; j++, ip += channels) {
-    out[op] = in[ip] * 2 + 1;
-    op = (op + 1) % nfft;
-  }
-}
-
-static void

 input_data_int16_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {
 input_data_int16_max (const guint8 * _in, gfloat * out, guint len,
     guint channels, gfloat max_value, guint op, guint nfft)
 {


@@ -739,54 +614,39 @@ input_data_int16_max (const guint8 * _in, gfloat * out, guint len,
 }
 
 static gboolean
 }
 
 static gboolean

-gst_spectrum_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+gst_spectrum_setup (GstAudioFilter * base, GstAudioInfo * info)

 {
   GstSpectrum *spectrum = GST_SPECTRUM (base);
 {
   GstSpectrum *spectrum = GST_SPECTRUM (base);

-  guint width = format->width / 8;
-  gboolean is_float = (format->type == GST_BUFTYPE_FLOAT);
-  /* max_value will be 0 when depth is 1,
-   * interpret -1 and 0 as -1 and +1 if that's the case. */
-  guint max_value = (1UL << (format->depth - 1)) - 1;

   gboolean multi_channel = spectrum->multi_channel;
   GstSpectrumInputData input_data = NULL;
 
   gboolean multi_channel = spectrum->multi_channel;
   GstSpectrumInputData input_data = NULL;
 

-  if (is_float) {
-    if (width == 4) {
+  switch (GST_AUDIO_INFO_FORMAT (info)) {
+    case GST_AUDIO_FORMAT_S16:
+      input_data =
+          multi_channel ? input_data_int16_max : input_data_mixed_int16_max;
+      break;
+    case GST_AUDIO_FORMAT_S24_3:
+      input_data =
+          multi_channel ? input_data_int24_max : input_data_mixed_int24_max;
+      break;
+    case GST_AUDIO_FORMAT_S32:
+      input_data =
+          multi_channel ? input_data_int32_max : input_data_mixed_int32_max;
+      break;
+    case GST_AUDIO_FORMAT_F32:

       input_data = multi_channel ? input_data_float : input_data_mixed_float;
       input_data = multi_channel ? input_data_float : input_data_mixed_float;

-    } else if (width == 8) {
+      break;
+    case GST_AUDIO_FORMAT_F64:

       input_data = multi_channel ? input_data_double : input_data_mixed_double;
       input_data = multi_channel ? input_data_double : input_data_mixed_double;

-    } else {
-      g_assert_not_reached ();
-    }
-  } else {
-    if (width == 4) {
-      if (max_value) {
-        input_data =
-            multi_channel ? input_data_int32_max : input_data_mixed_int32_max;
-      } else {
-        input_data = multi_channel ? input_data_int32 : input_data_mixed_int32;
-      }
-    } else if (width == 3) {
-      if (max_value) {
-        input_data =
-            multi_channel ? input_data_int24_max : input_data_mixed_int24_max;
-      } else {
-        input_data = multi_channel ? input_data_int24 : input_data_mixed_int24;
-      }
-    } else if (width == 2) {
-      if (max_value) {
-        input_data =
-            multi_channel ? input_data_int16_max : input_data_mixed_int16_max;
-      } else {
-        input_data = multi_channel ? input_data_int16 : input_data_mixed_int16;
-      }
-    } else {
+      break;
+    default:

       g_assert_not_reached ();
       g_assert_not_reached ();

-    }
+      break;

   }
   }

-

   spectrum->input_data = input_data;
   spectrum->input_data = input_data;

+

   gst_spectrum_reset_state (spectrum);
   gst_spectrum_reset_state (spectrum);

+

   return TRUE;
 }
 
   return TRUE;
 }
 


@@ -878,7 +738,7 @@ gst_spectrum_message_new (GstSpectrum * spectrum, GstClockTime timestamp,
     }
   } else {
     guint c;
     }
   } else {
     guint c;

-    guint channels = GST_AUDIO_FILTER (spectrum)->format.channels;
+    guint channels = GST_AUDIO_FILTER_CHANNELS (spectrum);

 
     if (spectrum->message_magnitude) {
       mcv = gst_spectrum_message_add_container (s, GST_TYPE_ARRAY, "magnitude");
 
     if (spectrum->message_magnitude) {
       mcv = gst_spectrum_message_add_container (s, GST_TYPE_ARRAY, "magnitude");


@@ -984,20 +844,19 @@ static GstFlowReturn
 gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
 {
   GstSpectrum *spectrum = GST_SPECTRUM (trans);
 gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
 {
   GstSpectrum *spectrum = GST_SPECTRUM (trans);

-  GstRingBufferSpec *format = &GST_AUDIO_FILTER (spectrum)->format;
-  guint rate = format->rate;
-  guint channels = format->channels;
+  guint rate = GST_AUDIO_FILTER_RATE (spectrum);
+  guint channels = GST_AUDIO_FILTER_CHANNELS (spectrum);
+  guint bps = GST_AUDIO_FILTER_BPS (spectrum);
+  guint bpf = GST_AUDIO_FILTER_BPF (spectrum);

   guint output_channels = spectrum->multi_channel ? channels : 1;
   guint c;
   guint output_channels = spectrum->multi_channel ? channels : 1;
   guint c;

-  guint width = format->width / 8;
-  gfloat max_value = (1UL << (format->depth - 1)) - 1;
+  gfloat max_value = (1UL << ((bps << 3) - 1)) - 1;

   guint bands = spectrum->bands;
   guint nfft = 2 * bands - 2;
   guint input_pos;
   gfloat *input;
   const guint8 *data, *mdata;
   gsize size;
   guint bands = spectrum->bands;
   guint nfft = 2 * bands - 2;
   guint input_pos;
   gfloat *input;
   const guint8 *data, *mdata;
   gsize size;

-  guint frame_size = width * channels;

   guint fft_todo, msg_todo, block_size;
   gboolean have_full_interval;
   GstSpectrumChannel *cd;
   guint fft_todo, msg_todo, block_size;
   gboolean have_full_interval;
   GstSpectrumChannel *cd;


@@ -1047,16 +906,16 @@ gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
   input_pos = spectrum->input_pos;
   input_data = spectrum->input_data;
 
   input_pos = spectrum->input_pos;
   input_data = spectrum->input_data;
 

-  while (size >= frame_size) {
+  while (size >= bpf) {

     /* run input_data for a chunk of data */
     fft_todo = nfft - (spectrum->num_frames % nfft);
     msg_todo = spectrum->frames_todo - spectrum->num_frames;
     GST_LOG_OBJECT (spectrum,
         "message frames todo: %u, fft frames todo: %u, input frames %u",
     /* run input_data for a chunk of data */
     fft_todo = nfft - (spectrum->num_frames % nfft);
     msg_todo = spectrum->frames_todo - spectrum->num_frames;
     GST_LOG_OBJECT (spectrum,
         "message frames todo: %u, fft frames todo: %u, input frames %u",

-        msg_todo, fft_todo, (size / frame_size));
+        msg_todo, fft_todo, (size / bpf));

     block_size = msg_todo;
     block_size = msg_todo;

-    if (block_size > (size / frame_size))
-      block_size = (size / frame_size);
+    if (block_size > (size / bpf))
+      block_size = (size / bpf);

     if (block_size > fft_todo)
       block_size = fft_todo;
 
     if (block_size > fft_todo)
       block_size = fft_todo;
 


@@ -1064,11 +923,11 @@ gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
       cd = &spectrum->channel_data[c];
       input = cd->input;
       /* Move the current frames into our ringbuffers */
       cd = &spectrum->channel_data[c];
       input = cd->input;
       /* Move the current frames into our ringbuffers */

-      input_data (data + c * width, input, block_size, channels, max_value,
+      input_data (data + c * bps, input, block_size, channels, max_value,

           input_pos, nfft);
     }
           input_pos, nfft);
     }

-    data += block_size * frame_size;
-    size -= block_size * frame_size;
+    data += block_size * bpf;
+    size -= block_size * bpf;

     input_pos = (input_pos + block_size) % nfft;
     spectrum->num_frames += block_size;
 
     input_pos = (input_pos + block_size) % nfft;
     spectrum->num_frames += block_size;
 

diff --git a/sys/oss/gstosshelper.c b/sys/oss/gstosshelper.c
index 6d7e6bd..639ee9d 100644 (file)
--- a/sys/oss/gstosshelper.c
+++ b/sys/oss/gstosshelper.c
@@ -169,53 +169,34 @@ static GstStructure *
 gst_oss_helper_get_format_structure (unsigned int format_bit)
 {
   GstStructure *structure;
 gst_oss_helper_get_format_structure (unsigned int format_bit)
 {
   GstStructure *structure;

-  int endianness;
-  gboolean sign;
-  int width;
+  const gchar *format;

 
   switch (format_bit) {
     case AFMT_U8:
 
   switch (format_bit) {
     case AFMT_U8:

-      endianness = 0;
-      sign = FALSE;
-      width = 8;
+      format = "U8";

       break;
     case AFMT_S16_LE:
       break;
     case AFMT_S16_LE:

-      endianness = G_LITTLE_ENDIAN;
-      sign = TRUE;
-      width = 16;
+      format = "S16_LE";

       break;
     case AFMT_S16_BE:
       break;
     case AFMT_S16_BE:

-      endianness = G_BIG_ENDIAN;
-      sign = TRUE;
-      width = 16;
+      format = "S16_BE";

       break;
     case AFMT_S8:
       break;
     case AFMT_S8:

-      endianness = 0;
-      sign = TRUE;
-      width = 8;
+      format = "S8";

       break;
     case AFMT_U16_LE:
       break;
     case AFMT_U16_LE:

-      endianness = G_LITTLE_ENDIAN;
-      sign = FALSE;
-      width = 16;
+      format = "U16_LE";

       break;
     case AFMT_U16_BE:
       break;
     case AFMT_U16_BE:

-      endianness = G_BIG_ENDIAN;
-      sign = FALSE;
-      width = 16;
+      format = "U16_BE";

       break;
     default:
       g_assert_not_reached ();
       return NULL;
   }
 
       break;
     default:
       g_assert_not_reached ();
       return NULL;
   }
 

-  structure = gst_structure_new ("audio/x-raw-int",
-      "width", G_TYPE_INT, width,
-      "depth", G_TYPE_INT, width, "signed", G_TYPE_BOOLEAN, sign, NULL);
-
-  if (endianness) {
-    gst_structure_set (structure, "endianness", G_TYPE_INT, endianness, NULL);
-  }
+  structure = gst_structure_new ("audio/x-raw",
+      "format", G_TYPE_STRING, format, NULL);

 
   return structure;
 }
 
   return structure;
 }

diff --git a/sys/oss/gstosssink.c b/sys/oss/gstosssink.c
index 9f9b137..6a8046d 100644 (file)
--- a/sys/oss/gstosssink.c
+++ b/sys/oss/gstosssink.c
@@ -110,20 +110,14 @@ enum
   PROP_DEVICE,
 };
 
   PROP_DEVICE,
 };
 

+#define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }"
+

 static GstStaticPadTemplate osssink_sink_factory =
 static GstStaticPadTemplate osssink_sink_factory =

-    GST_STATIC_PAD_TEMPLATE ("sink",
+GST_STATIC_PAD_TEMPLATE ("sink",

     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_PAD_SINK,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
-        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
-        "audio/x-raw-int, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) 8, "
-        "depth = (int) 8, "
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " FORMATS ", "

         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );
 
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );
 

diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c
index 33f585d..bdadbbc 100644 (file)
--- a/sys/oss/gstosssrc.c
+++ b/sys/oss/gstosssrc.c
@@ -101,25 +101,16 @@ static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
 static guint gst_oss_src_delay (GstAudioSrc * asrc);
 static void gst_oss_src_reset (GstAudioSrc * asrc);
 
 static guint gst_oss_src_delay (GstAudioSrc * asrc);
 static void gst_oss_src_reset (GstAudioSrc * asrc);
 

-
+#define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }"

 
 static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
 
 static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,

-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
-        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
-        "audio/x-raw-int, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) 8, "
-        "depth = (int) 8, "
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " FORMATS ", "

         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );
 
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
     );
 

-

 static void
 gst_oss_src_dispose (GObject * object)
 {
 static void
 gst_oss_src_dispose (GObject * object)
 {

diff --git a/tests/check/elements/audioinvert.c b/tests/check/elements/audioinvert.c
index 39ff80e..35d0615 100644 (file)
--- a/tests/check/elements/audioinvert.c
+++ b/tests/check/elements/audioinvert.c
@@ -37,31 +37,24 @@ GstPad *mysrcpad, *mysinkpad;
 
 
 #define INVERT_CAPS_STRING    \
 
 
 #define INVERT_CAPS_STRING    \

-    "audio/x-raw-int, "                 \
+    "audio/x-raw, "                     \
+    "format = (string) "GST_AUDIO_NE(S16)", "   \

     "channels = (int) 1, "              \
     "channels = (int) 1, "              \

-    "rate = (int) 44100, "              \
-    "endianness = (int) BYTE_ORDER, "   \
-    "width = (int) 16, "                \
-    "depth = (int) 16, "                \
-    "signed = (bool) TRUE"
+    "rate = (int) 44100"

 
 static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
 
 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")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (S16) ", "
+        "channels = (int) 1, " "rate = (int) [ 1,  MAX ]")

     );
 static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     );
 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")
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (S16) ", "
+        "channels = (int) 1, " "rate = (int) [ 1,  MAX ]")

     );
 
 static GstElement *
     );
 
 static GstElement *

diff --git a/tests/check/elements/level.c b/tests/check/elements/level.c
index 2bb4eb7..7969855 100644 (file)
--- a/tests/check/elements/level.c
+++ b/tests/check/elements/level.c
@@ -33,23 +33,16 @@ gboolean have_eos = FALSE;
 GstPad *mysrcpad, *mysinkpad;
 
 #define LEVEL_CAPS_TEMPLATE_STRING \
 GstPad *mysrcpad, *mysinkpad;
 
 #define LEVEL_CAPS_TEMPLATE_STRING \

-  "audio/x-raw-int, " \
+  "audio/x-raw, " \
+    "format = (string) { S8, "GST_AUDIO_NE(S16)" }, " \

     "rate = (int) [ 1, MAX ], " \
     "rate = (int) [ 1, MAX ], " \

-    "channels = (int) [ 1, 8 ], " \
-    "endianness = (int) BYTE_ORDER, " \
-    "width = (int) {8, 16}, " \
-    "depth = (int) {8, 16}, " \
-    "signed = (boolean) true"
+    "channels = (int) [ 1, 8 ]"

 
 #define LEVEL_CAPS_STRING \
 
 #define LEVEL_CAPS_STRING \

-  "audio/x-raw-int, " \
+  "audio/x-raw, " \
+    "format = (string) "GST_AUDIO_NE(S16)", " \

     "rate = (int) 1000, " \
     "rate = (int) 1000, " \

-    "channels = (int) 2, " \
-    "endianness = (int) BYTE_ORDER, " \
-    "width = (int) 16, " \
-    "depth = (int) 16, " \
-    "signed = (boolean) true"
-
+    "channels = (int) 2"

 
 static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
 
 static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,

diff --git a/tests/check/elements/rtp-payloading.c b/tests/check/elements/rtp-payloading.c
index b2160f4..9025ab4 100644 (file)
--- a/tests/check/elements/rtp-payloading.c
+++ b/tests/check/elements/rtp-payloading.c
@@ -600,7 +600,7 @@ GST_START_TEST (rtp_L16)
 {
   rtp_pipeline_test (rtp_L16_frame_data, rtp_L16_frame_data_size,
       rtp_L16_frame_count,
 {
   rtp_pipeline_test (rtp_L16_frame_data, rtp_L16_frame_data_size,
       rtp_L16_frame_count,

-      "audio/x-raw-int,endianess=4321,signed=true,width=16,depth=16,rate=1,channels=1",
+      "audio/x-raw,format=S16_BE,rate=1,channels=1",

       "rtpL16pay", "rtpL16depay", 0, 0, FALSE);
 }
 
       "rtpL16pay", "rtpL16depay", 0, 0, FALSE);
 }
 

diff --git a/tests/check/elements/rtpjitterbuffer.c b/tests/check/elements/rtpjitterbuffer.c
index a8eaa61..619253b 100644 (file)
--- a/tests/check/elements/rtpjitterbuffer.c
+++ b/tests/check/elements/rtpjitterbuffer.c
@@ -68,8 +68,8 @@ setup_jitterbuffer (gint num_buffers)
   GstBuffer *buffer;
   GstCaps *caps;
   /* a 20 sample audio block (2,5 ms) generated with
   GstBuffer *buffer;
   GstCaps *caps;
   /* a 20 sample audio block (2,5 ms) generated with

-   * gst-launch audiotestsrc wave=silence blocksize=40 num-buffers=3 ! 
-   *    "audio/x-raw-int,channels=1,rate=8000" ! mulawenc ! rtppcmupay !
+   * gst-launch audiotestsrc wave=silence blocksize=40 num-buffers=3 !
+   *    "audio/x-raw,channels=1,rate=8000" ! mulawenc ! rtppcmupay !

    *     fakesink dump=1
    */
   guint8 in[] = {               /* first 4 bytes are rtp-header, next 4 bytes are timestamp */
    *     fakesink dump=1
    */
   guint8 in[] = {               /* first 4 bytes are rtp-header, next 4 bytes are timestamp */

diff --git a/tests/examples/level/level-example.c b/tests/examples/level/level-example.c
index a85d751..1ce625f 100644 (file)
--- a/tests/examples/level/level-example.c
+++ b/tests/examples/level/level-example.c
@@ -87,7 +87,7 @@ main (int argc, char *argv[])
 
   gst_init (&argc, &argv);
 
 
   gst_init (&argc, &argv);
 

-  caps = gst_caps_from_string ("audio/x-raw-int,channels=2");
+  caps = gst_caps_from_string ("audio/x-raw,channels=2");

 
   pipeline = gst_pipeline_new (NULL);
   g_assert (pipeline);
 
   pipeline = gst_pipeline_new (NULL);
   g_assert (pipeline);

diff --git a/tests/examples/spectrum/spectrum-example.c b/tests/examples/spectrum/spectrum-example.c
index b552921..c843d0c 100644 (file)
--- a/tests/examples/spectrum/spectrum-example.c
+++ b/tests/examples/spectrum/spectrum-example.c
@@ -95,7 +95,7 @@ main (int argc, char *argv[])
 
   gst_bin_add_many (GST_BIN (bin), src, audioconvert, spectrum, sink, NULL);
 
 
   gst_bin_add_many (GST_BIN (bin), src, audioconvert, spectrum, sink, NULL);
 

-  caps = gst_caps_new_simple ("audio/x-raw-int",
+  caps = gst_caps_new_simple ("audio/x-raw",

       "rate", G_TYPE_INT, AUDIOFREQ, NULL);
 
   if (!gst_element_link (src, audioconvert) ||
       "rate", G_TYPE_INT, AUDIOFREQ, NULL);
 
   if (!gst_element_link (src, audioconvert) ||