rtp: Update for the new audio caps

diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c
index ca9f3c0..f1bed59 100644 (file)
--- a/gst/rtp/gstrtpL16depay.c
+++ b/gst/rtp/gstrtpL16depay.c
@@ -25,7 +25,6 @@
 #include <stdlib.h>
 
 #include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
 
 #include "gstrtpL16depay.h"
 #include "gstrtpchannels.h"
@@ -39,6 +38,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
         "format = (string) S16_BE, "
+        "layout = (string) interleaved, "
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
 
@@ -131,6 +131,7 @@ gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
   gboolean res;
   const gchar *channel_order;
   const GstRTPChannelOrder *order;
+  GstAudioInfo *info;
 
   rtpL16depay = GST_RTP_L16_DEPAY (depayload);
 
@@ -171,32 +172,32 @@ gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
   }
 
   depayload->clock_rate = clock_rate;
-  rtpL16depay->rate = clock_rate;
-  rtpL16depay->channels = channels;
 
-  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);
+  info = &rtpL16depay->info;
+  gst_audio_info_init (info);
+  info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
+  info->rate = clock_rate;
+  info->channels = channels;
+  info->bpf = (info->finfo->width / 8) * channels;
 
   /* add channel positions */
   channel_order = gst_structure_get_string (structure, "channel-order");
 
   order = gst_rtp_channels_get_by_order (channels, channel_order);
+  rtpL16depay->order = order;
   if (order) {
-    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
-        order->pos);
+    memcpy (info->position, order->pos,
+        sizeof (GstAudioChannelPosition) * channels);
+    gst_audio_channel_positions_to_valid_order (info->position, info->channels);
   } else {
-    GstAudioChannelPosition *pos;
-
     GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
         (NULL), ("Unknown channel order '%s' for %d channels",
             GST_STR_NULL (channel_order), channels));
     /* create default NONE layout */
-    pos = gst_rtp_channels_create_default (channels);
-    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
-    g_free (pos);
+    gst_rtp_channels_create_default (channels, info->position);
   }
 
+  srccaps = gst_audio_info_to_caps (info);
   res = gst_pad_set_caps (depayload->srcpad, srccaps);
   gst_caps_unref (srccaps);
 
@@ -237,6 +238,14 @@ gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
   }
 
+  outbuf = gst_buffer_make_writable (outbuf);
+  if (rtpL16depay->order &&
+      !gst_audio_buffer_reorder_channels (outbuf,
+          rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
+          rtpL16depay->info.position, rtpL16depay->order->pos)) {
+    goto reorder_failed;
+  }
+
   gst_rtp_buffer_unmap (&rtp);
 
   return outbuf;
@@ -249,6 +258,13 @@ empty_packet:
     gst_rtp_buffer_unmap (&rtp);
     return NULL;
   }
+reorder_failed:
+  {
+    GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
+        ("Channel reordering failed."), (NULL));
+    gst_rtp_buffer_unmap (&rtp);
+    return NULL;
+  }
 }
 
 gboolean

diff --git a/gst/rtp/gstrtpL16depay.h b/gst/rtp/gstrtpL16depay.h
index 0fd9506..b5b7c93 100644 (file)
--- a/gst/rtp/gstrtpL16depay.h
+++ b/gst/rtp/gstrtpL16depay.h
@@ -22,6 +22,9 @@
 
 #include <gst/gst.h>
 #include <gst/rtp/gstrtpbasedepayload.h>
+#include <gst/audio/audio.h>
+
+#include "gstrtpchannels.h"
 
 G_BEGIN_DECLS
 
@@ -45,8 +48,8 @@ struct _GstRtpL16Depay
 {
   GstRTPBaseDepayload depayload;
 
-  guint rate;
-  guint channels;
+  GstAudioInfo info;
+  const GstRTPChannelOrder *order;
 };
 
 /* Standard definition defining a class for this element. */

diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c
index cf5a647..af98ba9 100644 (file)
--- a/gst/rtp/gstrtpL16pay.c
+++ b/gst/rtp/gstrtpL16pay.c
@@ -24,7 +24,6 @@
 #include <string.h>
 
 #include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
 #include <gst/rtp/gstrtpbuffer.h>
 
 #include "gstrtpL16pay.h"
@@ -39,6 +38,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw, "
         "format = (string) S16_BE, "
+        "layout = (string) interleaved, "
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
     );
 
@@ -68,6 +68,9 @@ static gboolean gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload,
     GstCaps * caps);
 static GstCaps *gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload,
     GstPad * pad, GstCaps * filter);
+static GstFlowReturn
+gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
+    GstBuffer * buffer);
 
 #define gst_rtp_L16_pay_parent_class parent_class
 G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
@@ -83,6 +86,7 @@ gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
 
   gstrtpbasepayload_class->set_caps = gst_rtp_L16_pay_setcaps;
   gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps;
+  gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
 
   gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
@@ -113,72 +117,54 @@ static gboolean
 gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
 {
   GstRtpL16Pay *rtpL16pay;
-  GstStructure *structure;
-  gint channels, rate;
   gboolean res;
   gchar *params;
-  GstAudioChannelPosition *pos;
+  GstAudioInfo *info;
   const GstRTPChannelOrder *order;
   GstRTPBaseAudioPayload *rtpbaseaudiopayload;
 
   rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
   rtpL16pay = GST_RTP_L16_PAY (basepayload);
 
-  structure = gst_caps_get_structure (caps, 0);
+  info = &rtpL16pay->info;
+  gst_audio_info_init (info);
+  if (!gst_audio_info_from_caps (info, caps))
+    goto invalid_caps;
 
-  /* first parse input caps */
-  if (!gst_structure_get_int (structure, "rate", &rate))
-    goto no_rate;
+  order = gst_rtp_channels_get_by_pos (info->channels, info->position);
+  rtpL16pay->order = order;
 
-  if (!gst_structure_get_int (structure, "channels", &channels))
-    goto no_channels;
+  gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16",
+      info->rate);
+  params = g_strdup_printf ("%d", info->channels);
 
-  /* get the channel order */
-  pos = gst_audio_get_channel_positions (structure);
-  if (pos)
-    order = gst_rtp_channels_get_by_pos (channels, pos);
-  else
-    order = NULL;
-
-  gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16", rate);
-  params = g_strdup_printf ("%d", channels);
-
-  if (!order && channels > 2) {
+  if (!order && info->channels > 2) {
     GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
-        (NULL), ("Unknown channel order for %d channels", channels));
+        (NULL), ("Unknown channel order for %d channels", info->channels));
   }
 
   if (order && order->name) {
     res = gst_rtp_base_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
-        channels, "channel-order", G_TYPE_STRING, order->name, NULL);
+        info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
   } else {
     res = gst_rtp_base_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
-        channels, NULL);
+        info->channels, NULL);
   }
 
   g_free (params);
-  g_free (pos);
-
-  rtpL16pay->rate = rate;
-  rtpL16pay->channels = channels;
 
   /* octet-per-sample is 2 * channels for L16 */
   gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
-      2 * rtpL16pay->channels);
+      2 * info->channels);
 
   return res;
 
   /* ERRORS */
-no_rate:
-  {
-    GST_DEBUG_OBJECT (rtpL16pay, "no rate given");
-    return FALSE;
-  }
-no_channels:
+invalid_caps:
   {
-    GST_DEBUG_OBJECT (rtpL16pay, "no channels given");
+    GST_DEBUG_OBJECT (rtpL16pay, "invalid caps");
     return FALSE;
   }
 }
@@ -232,6 +218,26 @@ gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
   return caps;
 }
 
+static GstFlowReturn
+gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
+    GstBuffer * buffer)
+{
+  GstRtpL16Pay *rtpL16pay;
+
+  rtpL16pay = GST_RTP_L16_PAY (basepayload);
+  buffer = gst_buffer_make_writable (buffer);
+
+  if (rtpL16pay->order &&
+      !gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format,
+          rtpL16pay->info.channels, rtpL16pay->info.position,
+          rtpL16pay->order->pos)) {
+    return GST_FLOW_ERROR;
+  }
+
+  return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
+      buffer);
+}
+
 gboolean
 gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
 {

diff --git a/gst/rtp/gstrtpL16pay.h b/gst/rtp/gstrtpL16pay.h
index b24138e..515a512 100644 (file)
--- a/gst/rtp/gstrtpL16pay.h
+++ b/gst/rtp/gstrtpL16pay.h
@@ -23,6 +23,8 @@
 #include <gst/gst.h>
 #include <gst/rtp/gstrtpbaseaudiopayload.h>
 
+#include "gstrtpchannels.h"
+
 G_BEGIN_DECLS
 
 #define GST_TYPE_RTP_L16_PAY \
@@ -43,8 +45,8 @@ struct _GstRtpL16Pay
 {
   GstRTPBaseAudioPayload payload;
 
-  gint rate;
-  gint channels;
+  GstAudioInfo info;
+  const GstRTPChannelOrder *order;
 };
 
 struct _GstRtpL16PayClass

diff --git a/gst/rtp/gstrtpchannels.c b/gst/rtp/gstrtpchannels.c
index 2ce4280..2c56ec2 100644 (file)
--- a/gst/rtp/gstrtpchannels.c
+++ b/gst/rtp/gstrtpchannels.c
@@ -33,14 +33,16 @@ static gboolean
 check_channels (const GstRTPChannelOrder * order,
     const GstAudioChannelPosition * pos)
 {
-  gint i;
+  gint i, j;
   gboolean res = TRUE;
 
   for (i = 0; i < order->channels; i++) {
-    if (order->pos[i] != pos[i]) {
-      res = FALSE;
-      break;
+    for (j = 0; j < order->channels; j++) {
+      if (order->pos[j] == pos[i])
+        break;
     }
+    if (j == order->channels)
+      return FALSE;
   }
   return res;
 }
@@ -150,18 +152,13 @@ gst_rtp_channels_get_by_index (gint channels, guint idx)
  * Returns: a #GstAudioChannelPosition with all the channel position info set to
  * #GST_AUDIO_CHANNEL_POSITION_NONE.
  */
-GstAudioChannelPosition *
-gst_rtp_channels_create_default (gint channels)
+void
+gst_rtp_channels_create_default (gint channels, GstAudioChannelPosition * posn)
 {
   gint i;
-  GstAudioChannelPosition *posn;
 
-  g_return_val_if_fail (channels > 0, NULL);
-
-  posn = g_new (GstAudioChannelPosition, channels);
+  g_return_if_fail (channels > 0);
 
   for (i = 0; i < channels; i++)
     posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
-
-  return posn;
 }

diff --git a/gst/rtp/gstrtpchannels.h b/gst/rtp/gstrtpchannels.h
index b99a200..0a1b3a5 100644 (file)
--- a/gst/rtp/gstrtpchannels.h
+++ b/gst/rtp/gstrtpchannels.h
@@ -21,7 +21,9 @@
 #include <stdlib.h>
 
 #include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
+
+#ifndef __GST_RTP_CHANNELS_H__
+#define __GST_RTP_CHANNELS_H__
 
 typedef struct
 {
@@ -41,14 +43,14 @@ static const GstAudioChannelPosition pos_4_2[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE
+  GST_AUDIO_CHANNEL_POSITION_LFE1
 };
 
 static const GstAudioChannelPosition pos_4_3[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE
+  GST_AUDIO_CHANNEL_POSITION_LFE1
 };
 
 static const GstAudioChannelPosition pos_5_1[] = {
@@ -65,14 +67,14 @@ static const GstAudioChannelPosition pos_6_1[] = {
   GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
   GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE
+  GST_AUDIO_CHANNEL_POSITION_LFE1
 };
 
 static const GstAudioChannelPosition pos_6_2[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE,
+  GST_AUDIO_CHANNEL_POSITION_LFE1,
   GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
   GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
 };
@@ -81,7 +83,7 @@ static const GstAudioChannelPosition pos_8_1[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE,
+  GST_AUDIO_CHANNEL_POSITION_LFE1,
   GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
   GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
@@ -92,7 +94,7 @@ static const GstAudioChannelPosition pos_8_2[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE,
+  GST_AUDIO_CHANNEL_POSITION_LFE1,
   GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
   GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
@@ -103,7 +105,7 @@ static const GstAudioChannelPosition pos_8_3[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-  GST_AUDIO_CHANNEL_POSITION_LFE,
+  GST_AUDIO_CHANNEL_POSITION_LFE1,
   GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
   GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
@@ -111,7 +113,7 @@ static const GstAudioChannelPosition pos_8_3[] = {
 };
 
 static const GstAudioChannelPosition pos_def_1[] = {
-  GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
+  GST_AUDIO_CHANNEL_POSITION_MONO
 };
 
 static const GstAudioChannelPosition pos_def_2[] = {
@@ -129,7 +131,7 @@ static const GstAudioChannelPosition pos_def_4[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-  GST_AUDIO_CHANNEL_POSITION_LFE
+  GST_AUDIO_CHANNEL_POSITION_LFE1
 };
 
 static const GstAudioChannelPosition pos_def_5[] = {
@@ -146,7 +148,7 @@ static const GstAudioChannelPosition pos_def_6[] = {
   GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
   GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
   GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
-  GST_AUDIO_CHANNEL_POSITION_LFE
+  GST_AUDIO_CHANNEL_POSITION_LFE1
 };
 
 static const GstRTPChannelOrder channel_orders[] =
@@ -183,4 +185,6 @@ const GstRTPChannelOrder *   gst_rtp_channels_get_by_order   (gint channels,
                                                               const gchar *order);
 const GstRTPChannelOrder *   gst_rtp_channels_get_by_index   (gint channels, guint idx);
 
-GstAudioChannelPosition *    gst_rtp_channels_create_default (gint channels);
+void                         gst_rtp_channels_create_default (gint channels, GstAudioChannelPosition *pos);
+
+#endif /* __GST_RTP_CHANNELS_H__ */

diff --git a/gst/rtp/gstrtpg722depay.c b/gst/rtp/gstrtpg722depay.c
index b3c66e3..e197e4d 100644 (file)
--- a/gst/rtp/gstrtpg722depay.c
+++ b/gst/rtp/gstrtpg722depay.c
@@ -25,7 +25,6 @@
 #include <stdlib.h>
 
 #include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
 
 #include "gstrtpg722depay.h"
 #include "gstrtpchannels.h"
@@ -127,8 +126,10 @@ gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
   gint channels;
   GstCaps *srccaps;
   gboolean res;
+#if 0
   const gchar *channel_order;
   const GstRTPChannelOrder *order;
+#endif
 
   rtpg722depay = GST_RTP_G722_DEPAY (depayload);
 
@@ -179,6 +180,8 @@ gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
   srccaps = gst_caps_new_simple ("audio/G722",
       "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
 
+  /* FIXME: Do something with the channel order */
+#if 0
   /* add channel positions */
   channel_order = gst_structure_get_string (structure, "channel-order");
 
@@ -197,6 +200,7 @@ gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
     gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
     g_free (pos);
   }
+#endif
 
   res = gst_pad_set_caps (depayload->srcpad, srccaps);
   gst_caps_unref (srccaps);

diff --git a/gst/rtp/gstrtpg722pay.c b/gst/rtp/gstrtpg722pay.c
index a565f38..986b84c 100644 (file)
--- a/gst/rtp/gstrtpg722pay.c
+++ b/gst/rtp/gstrtpg722pay.c
@@ -24,7 +24,6 @@
 #include <string.h>
 
 #include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
 #include <gst/rtp/gstrtpbuffer.h>
 
 #include "gstrtpg722pay.h"
@@ -105,8 +104,10 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
   gint rate, channels, clock_rate;
   gboolean res;
   gchar *params;
+#if 0
   GstAudioChannelPosition *pos;
   const GstRTPChannelOrder *order;
+#endif
   GstRTPBaseAudioPayload *rtpbaseaudiopayload;
 
   rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
@@ -121,12 +122,15 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
   if (!gst_structure_get_int (structure, "channels", &channels))
     goto no_channels;
 
+  /* FIXME: Do something with the channel positions */
+#if 0
   /* get the channel order */
   pos = gst_audio_get_channel_positions (structure);
   if (pos)
     order = gst_rtp_channels_get_by_pos (channels, pos);
   else
     order = NULL;
+#endif
 
   /* Clock rate is always 8000 Hz for G722 according to
    * RFC 3551 although the sampling rate is 16000 Hz */
@@ -136,6 +140,7 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
       clock_rate);
   params = g_strdup_printf ("%d", channels);
 
+#if 0
   if (!order && channels > 2) {
     GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
         (NULL), ("Unknown channel order for %d channels", channels));
@@ -146,13 +151,18 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
         channels, "channel-order", G_TYPE_STRING, order->name, NULL);
   } else {
+#endif
     res = gst_rtp_base_payload_set_outcaps (basepayload,
         "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
         channels, NULL);
+#if 0
   }
+#endif
 
   g_free (params);
+#if 0
   g_free (pos);
+#endif
 
   rtpg722pay->rate = rate;
   rtpg722pay->channels = channels;

diff --git a/gst/rtp/gstrtpvrawpay.c b/gst/rtp/gstrtpvrawpay.c
index a463ef8..214a307 100644 (file)
--- a/gst/rtp/gstrtpvrawpay.c
+++ b/gst/rtp/gstrtpvrawpay.c
@@ -271,7 +271,7 @@ gst_rtp_vraw_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
   width = GST_VIDEO_INFO_WIDTH (&rtpvrawpay->vinfo);
   height = GST_VIDEO_INFO_HEIGHT (&rtpvrawpay->vinfo);
 
-  interlaced = !!(rtpvrawpay->vinfo.flags & GST_VIDEO_FLAG_INTERLACED);
+  interlaced = ! !(rtpvrawpay->vinfo.flags & GST_VIDEO_FLAG_INTERLACED);
 
   /* start with line 0, offset 0 */
   for (field = 0; field < 1 + interlaced; field++) {