[farsight2/master] Remove unused parameter in internal function
Olivier Crête
olivier.crete at collabora.co.uk
Tue Dec 23 15:24:10 PST 2008
---
gst/fsrtpconference/fs-rtp-session.c | 39 ++++---------------------------
gst/fsrtpconference/fs-rtp-substream.c | 7 +----
gst/fsrtpconference/fs-rtp-substream.h | 3 +-
3 files changed, 8 insertions(+), 41 deletions(-)
diff --git a/gst/fsrtpconference/fs-rtp-session.c b/gst/fsrtpconference/fs-rtp-session.c
index 2af6afc..374258a 100644
--- a/gst/fsrtpconference/fs-rtp-session.c
+++ b/gst/fsrtpconference/fs-rtp-session.c
@@ -1764,34 +1764,6 @@ fs_rtp_session_get_stream_by_ssrc (FsRtpSession *self,
return stream;
}
-/**
- * fs_rtp_session_verify_substream
- *
- * Verifies that a substream still has the right codec and resets it if its not
- *
- * MUST hold the FsRtpSession lock while calling it
- */
-
-static void
-fs_rtp_session_verify_substream_locked (FsRtpSession *session,
- FsRtpStream *stream,
- FsRtpSubStream *substream)
-{
- FsCodec *codec = NULL;
- guint pt;
-
- g_object_get (substream, "pt", &pt, NULL);
-
- codec = fs_rtp_session_get_recv_codec_locked (session, pt, stream, NULL,
- NULL);
-
- if (!codec)
- return;
-
- fs_rtp_sub_stream_verify_codec_locked (substream, codec);
-
- fs_codec_destroy (codec);
-}
/**
* fs_rtp_session_verify_recv_codecs
@@ -1811,7 +1783,7 @@ fs_rtp_session_verify_recv_codecs (FsRtpSession *session)
for (item = g_list_first (session->priv->free_substreams);
item;
item = g_list_next (item))
- fs_rtp_session_verify_substream_locked (session, NULL, item->data);
+ fs_rtp_sub_stream_verify_codec_locked (item->data);
for (item = g_list_first (session->priv->streams);
item;
@@ -1822,7 +1794,8 @@ fs_rtp_session_verify_recv_codecs (FsRtpSession *session)
for (item2 = g_list_first (stream->substreams);
item2;
item2 = g_list_next (item2))
- fs_rtp_session_verify_substream_locked (session, stream, item2->data);
+ fs_rtp_sub_stream_verify_codec_locked (item2->data);
+
}
FS_RTP_SESSION_UNLOCK (session);
@@ -3129,7 +3102,7 @@ fs_rtp_session_associate_free_substreams (FsRtpSession *session,
if (fs_rtp_stream_add_substream (stream, substream, &error))
{
- fs_rtp_session_verify_substream_locked (session, stream, substream);
+ fs_rtp_sub_stream_verify_codec_locked (substream);
GST_DEBUG ("Associated SSRC %x in session %u", ssrc, session->id);
}
else
@@ -3241,9 +3214,7 @@ _substream_no_rtcp_timedout_cb (FsRtpSubStream *substream,
if (fs_rtp_stream_add_substream (
g_list_first (session->priv->streams)->data,
substream, &error))
- fs_rtp_session_verify_substream_locked (session,
- g_list_first (session->priv->streams)->data,
- substream);
+ fs_rtp_sub_stream_verify_codec_locked (substream);
else
fs_session_emit_error (FS_SESSION (session),
error ? error->code : FS_ERROR_INTERNAL,
diff --git a/gst/fsrtpconference/fs-rtp-substream.c b/gst/fsrtpconference/fs-rtp-substream.c
index 1c11dbd..e4628b0 100644
--- a/gst/fsrtpconference/fs-rtp-substream.c
+++ b/gst/fsrtpconference/fs-rtp-substream.c
@@ -1158,18 +1158,15 @@ fs_rtp_sub_stream_add_probe_locked (FsRtpSubStream *substream)
/**
* fs_rtp_sub_stream_verify_codec_locked:
* @substream: A #FsRtpSubStream
- * @codec: The new fscodec (the substream is invalidated if it not using this
- * codec). You can pass NULL to match any codec.
*
* This function will start the process that invalidates the codec
- * for this rtpbin, if it doesnt match the passed codec
+ * for this rtpbin.
*
* You must hold the session lock to call it.
*/
void
-fs_rtp_sub_stream_verify_codec_locked (FsRtpSubStream *substream,
- const FsCodec *codec)
+fs_rtp_sub_stream_verify_codec_locked (FsRtpSubStream *substream)
{
fs_rtp_sub_stream_add_probe_locked (substream);
diff --git a/gst/fsrtpconference/fs-rtp-substream.h b/gst/fsrtpconference/fs-rtp-substream.h
index 36b18fd..13a84b6 100644
--- a/gst/fsrtpconference/fs-rtp-substream.h
+++ b/gst/fsrtpconference/fs-rtp-substream.h
@@ -91,8 +91,7 @@ gboolean fs_rtp_sub_stream_add_output_ghostpad_locked (
FsRtpSubStream *substream,
GError **error);
-void fs_rtp_sub_stream_verify_codec_locked (FsRtpSubStream *substream,
- const FsCodec *codec);
+void fs_rtp_sub_stream_verify_codec_locked (FsRtpSubStream *substream);
G_END_DECLS
--
1.5.6.5
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