From b891ea8e813463c9e54113c3c02e8c8ffb3d2f9a Mon Sep 17 00:00:00 2001
From: kapil <kapil@fluendo.com>
Date: Mon, 31 Mar 2008 16:24:42 +0000
Subject: ext/gsm/gstgsmdec.*: Increase the allowed samplerates for the ms-gsm
 format.

Original commit message from CVS:
Patch by: kapil <kapil at fluendo dot com>
* ext/gsm/gstgsmdec.c: (gst_gsmdec_sink_setcaps),
(gst_gsmdec_chain):
* ext/gsm/gstgsmdec.h:
Increase the allowed samplerates for the ms-gsm format.
Fixes #481354.
---
 ChangeLog           | 10 ++++++++++
 ext/gsm/gstgsmdec.c | 49 ++++++++++++++++++++++++++++++++++---------------
 ext/gsm/gstgsmdec.h |  4 ++++
 3 files changed, 48 insertions(+), 15 deletions(-)

diff --git a/ChangeLog b/ChangeLog
index 1446b239..0e2ee1b1 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,13 @@
+2008-03-31  Wim Taymans  <wim.taymans@collabora.co.uk>
+
+	Patch by: kapil <kapil at fluendo dot com>
+
+	* ext/gsm/gstgsmdec.c: (gst_gsmdec_sink_setcaps),
+	(gst_gsmdec_chain):
+	* ext/gsm/gstgsmdec.h:
+	Increase the allowed samplerates for the ms-gsm format.
+	Fixes #481354.
+
 2008-03-30  Tim-Philipp Müller  <tim at centricular dot net>
 
 	Patch by: Jan de Groot <jan at jgc homeip net>
diff --git a/ext/gsm/gstgsmdec.c b/ext/gsm/gstgsmdec.c
index c20c72a8..35dd1774 100644
--- a/ext/gsm/gstgsmdec.c
+++ b/ext/gsm/gstgsmdec.c
@@ -87,12 +87,14 @@ gst_gsmdec_get_type (void)
   return gsmdec_type;
 }
 
+#define ENCODED_SAMPLES	160
+
 static GstStaticPadTemplate gsmdec_sink_template =
     GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-gsm, rate = (int) 8000, channels = (int) 1; "
-        "audio/ms-gsm, rate = (int) 8000, channels = (int) 1")
+        "audio/ms-gsm, rate = (int) [1, MAX], channels = (int) 1")
     );
 
 static GstStaticPadTemplate gsmdec_src_template =
@@ -103,7 +105,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
         "endianness = (int) BYTE_ORDER, "
         "signed = (boolean) true, "
         "width = (int) 16, "
-        "depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
+        "depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
     );
 
 static void
@@ -175,6 +177,7 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
   GstGSMDec *gsmdec;
   GstCaps *srccaps;
   GstStructure *s;
+  gboolean ret = FALSE;
 
   gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
 
@@ -190,25 +193,41 @@ gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
   else
     goto wrong_caps;
 
+  if (!gst_structure_get_int (s, "rate", &gsmdec->rate)) {
+    GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
+    goto beach;
+  }
+
   /* MSGSM needs different framing */
   gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
 
-  /* we only have one possible source caps, which is the same as our template. */
-  srccaps = gst_static_pad_template_get_caps (&gsmdec_src_template);
+  gsmdec->duration = gst_util_uint64_scale (ENCODED_SAMPLES,
+      GST_SECOND, gsmdec->rate);
 
-  gst_pad_set_caps (gsmdec->srcpad, srccaps);
+  /* Setting up src caps based on the input sample rate. */
+  srccaps = gst_caps_new_simple ("audio/x-raw-int",
+      "endianness", G_TYPE_INT, BYTE_ORDER,
+      "signed", G_TYPE_BOOLEAN, TRUE,
+      "width", G_TYPE_INT, 16,
+      "depth", G_TYPE_INT, 16,
+      "rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL);
 
+  ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
+
+  gst_caps_unref (srccaps);
   gst_object_unref (gsmdec);
 
-  return TRUE;
+  return ret;
 
   /* ERRORS */
 wrong_caps:
-  {
-    GST_ERROR_OBJECT (gsmdec, "invalid caps received");
-    gst_object_unref (gsmdec);
-    return FALSE;
-  }
+
+  GST_ERROR_OBJECT (gsmdec, "invalid caps received");
+
+beach:
+  gst_object_unref (gsmdec);
+
+  return ret;
 }
 
 static gboolean
@@ -283,7 +302,7 @@ gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
     GstBuffer *outbuf;
 
     /* always the same amount of output samples */
-    outbuf = gst_buffer_new_and_alloc (160 * sizeof (gsm_signal));
+    outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
 
     /* If we are not given any timestamp, interpolate from last seen
      * timestamp (if any). */
@@ -294,13 +313,13 @@ gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
 
     /* interpolate in the next run */
     if (timestamp != GST_CLOCK_TIME_NONE)
-      gsmdec->next_ts = timestamp + (20 * GST_MSECOND);
+      gsmdec->next_ts = timestamp + gsmdec->duration;
     timestamp = GST_CLOCK_TIME_NONE;
 
-    GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
+    GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
     GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
     if (gsmdec->next_of != -1)
-      gsmdec->next_of += 160;
+      gsmdec->next_of += ENCODED_SAMPLES;
     GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
 
     gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
diff --git a/ext/gsm/gstgsmdec.h b/ext/gsm/gstgsmdec.h
index cc633d86..0013aa47 100644
--- a/ext/gsm/gstgsmdec.h
+++ b/ext/gsm/gstgsmdec.h
@@ -60,6 +60,10 @@ struct _GstGSMDec
   GstAdapter *adapter;
 
   GstSegment segment;
+
+  gint rate;
+
+  GstClockTime duration;
 };
 
 struct _GstGSMDecClass
-- 
cgit v1.2.1