ext/mad/gstid3tag.c: move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"

Original commit message from CVS:
* ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
* gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"

* gst/auparse/gstauparse.c :
- add code (commented for now) to support audio/x-adpcm on src pad
(we have no decoder for those layout yet)

* gst/cdxaparse/gstcdxaparse.c :
* gst/cdxaparse/gstcdxaparse.h :
- partial rewrite using RiffRead (ripped iain's wavparse code)

* gst/rtp/gstrtpL16enc.c : typo
* gst/rtp/gstrtpgsmenc.c : typo

diff --git a/ChangeLog b/ChangeLog
index 7e72d4e..43a4836 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,19 @@
+2004-05-13  Stephane Loeuillet <stephane.loeuillet@tiscali.fr>
+
+       * ext/mad/gstid3tag.c : move from "Codec/(Dem/M)uxer" to "Codec/(Dem/M)uxer/Audio"
+       * gst/wavenc/gstwavenc.c : move from "Codec/Encoder/Audio" to "Codec/Muxer/Audio"
+
+       * gst/auparse/gstauparse.c :
+         - add code (commented for now) to support audio/x-adpcm on src pad
+           (we have no decoder for those layout yet)
+
+       * gst/cdxaparse/gstcdxaparse.c :
+       * gst/cdxaparse/gstcdxaparse.h :
+         - partial rewrite using RiffRead (ripped iain's wavparse code)
+
+       * gst/rtp/gstrtpL16enc.c : typo
+       * gst/rtp/gstrtpgsmenc.c : typo
+
 2004-05-13  Benjamin Otte  <otte@gnome.org>
 
        * configure.ac:

diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c
index 5e16d3e..f3dacd0 100644 (file)
--- a/gst/auparse/gstauparse.c
+++ b/gst/auparse/gstauparse.c
@@ -51,11 +51,10 @@ static GstStaticPadTemplate gst_auparse_src_template =
     GST_PAD_SOMETIMES,          /* FIXME: spider */
     GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "       /* 24-bit PCM is barely supported by gstreamer actually */
         GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS "; "  /* 64-bit float is barely supported by gstreamer actually */
-        "audio/x-alaw, "
-        "rate = (int) [ 8000, 192000 ], "
-        "channels = (int) [ 1, 2 ]; "
-        "audio/x-mulaw, "
-        "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]")
+        "audio/x-alaw, " "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]; " "audio/x-mulaw, " "rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]"        /*"; "
+                                                                                                                                                                                   "audio/x-adpcm, "
+                                                                           "layout = (string) { g721, g722, g723_3, g723_5 }" */ )
+                                                                        /* Nothing to decode those ADPCM streams for now */
     );
 
 /* AuParse signals and args */
@@ -314,6 +313,11 @@ Samples :
           "width", G_TYPE_INT, depth,
           "endianness", G_TYPE_INT,
           auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, NULL);
+/*
+    } else if (layout) {
+      tempcaps = gst_caps_new_simple ("audio/x-adpcm",
+          "layout", G_TYPE_STRING, layout, NULL);
+*/
     } else {
       tempcaps = gst_caps_new_simple ("audio/x-raw-int",
           "endianness", G_TYPE_INT,

diff --git a/gst/rtp/gstrtpL16enc.c b/gst/rtp/gstrtpL16enc.c
index 373d2d6..1e3088e 100644 (file)
--- a/gst/rtp/gstrtpL16enc.c
+++ b/gst/rtp/gstrtpL16enc.c
@@ -28,7 +28,7 @@
 static GstElementDetails gst_rtpL16enc_details = {
   "RTP RAW Audio Encoder",
   "Codec/Encoder/Network",
-  "Encodes Raw Audio into an RTP packet",
+  "Encodes Raw Audio into a RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
 

diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index 373d2d6..1e3088e 100644 (file)
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -28,7 +28,7 @@
 static GstElementDetails gst_rtpL16enc_details = {
   "RTP RAW Audio Encoder",
   "Codec/Encoder/Network",
-  "Encodes Raw Audio into an RTP packet",
+  "Encodes Raw Audio into a RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
 

diff --git a/gst/rtp/gstrtpgsmenc.c b/gst/rtp/gstrtpgsmenc.c
index f95cc7e..ae441e1 100644 (file)
--- a/gst/rtp/gstrtpgsmenc.c
+++ b/gst/rtp/gstrtpgsmenc.c
@@ -29,7 +29,7 @@
 static GstElementDetails gst_rtpgsmenc_details = {
   "RTP GSM Audio Encoder",
   "Codec/Encoder/Network",
-  "Encodes GSM audio into an RTP packet",
+  "Encodes GSM audio into a RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
 

diff --git a/gst/rtp/gstrtpgsmpay.c b/gst/rtp/gstrtpgsmpay.c
index f95cc7e..ae441e1 100644 (file)
--- a/gst/rtp/gstrtpgsmpay.c
+++ b/gst/rtp/gstrtpgsmpay.c
@@ -29,7 +29,7 @@
 static GstElementDetails gst_rtpgsmenc_details = {
   "RTP GSM Audio Encoder",
   "Codec/Encoder/Network",
-  "Encodes GSM audio into an RTP packet",
+  "Encodes GSM audio into a RTP packet",
   "Zeeshan Ali <zak147@yahoo.com>"
 };
 

diff --git a/gst/wavenc/gstwavenc.c b/gst/wavenc/gstwavenc.c
index 99e34ce..7c3e9de 100644 (file)
--- a/gst/wavenc/gstwavenc.c
+++ b/gst/wavenc/gstwavenc.c
@@ -75,7 +75,7 @@ struct wave_header
 
 static GstElementDetails gst_wavenc_details =
 GST_ELEMENT_DETAILS ("WAV encoder",
-    "Codec/Encoder/Audio",
+    "Codec/Muxer/Audio",
     "Encode raw audio into WAV",
     "Iain Holmes <iain@prettypeople.org>");