Remove trailing whitespace to make git happy

7 files changed, 31 insertions, 31 deletions

diff --git a/src/debug/local-play.c b/src/debug/local-play.c
index 6658652..8dbb90d 100644
--- a/src/debug/local-play.c
+++ b/src/debug/local-play.c
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
diff --git a/src/debug/local-play.h b/src/debug/local-play.h
index ec6a24a..131a2af 100644
--- a/src/debug/local-play.h
+++ b/src/debug/local-play.h
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
diff --git a/src/encode.c b/src/encode.c
index 7dca6f7..88d6707 100644
--- a/src/encode.c
+++ b/src/encode.c
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
@@ -68,7 +68,7 @@ on_decodebin_pad_added (GstElement *decodebin,
   /* BUG: We don't try to fetch a compatible pad for raw audio because
    * that somehow always fails to link. Transmageddon does the same. */
   if (!g_str_has_prefix (name, "audio/x-raw"))
-    /* If we successfully requested a compatible sink pad in 
+    /* If we successfully requested a compatible sink pad in
      * "autoplug-continue", we can fetch that here. */
     sinkpad = gst_element_get_compatible_pad (encodebin, srcpad, NULL);
 
@@ -147,7 +147,7 @@ stp_encode_from_msg (TranscodeServerCtx *ctx)
   GstEncodingProfile *profile;
 
   g_debug ("Constructing pipeline\n");
-  
+
   src = gst_element_factory_make ("appsrc", "src");
   g_object_set (src, "is-live", TRUE,
                 "emit-signals", FALSE,
@@ -155,7 +155,7 @@ stp_encode_from_msg (TranscodeServerCtx *ctx)
   ctx->appsrc = src;
 
   decodebin = gst_element_factory_make ("decodebin", "decodebin");
-  
+
   gst_bin_add_many (GST_BIN (ctx->pipeline), src, decodebin, NULL);
   gst_element_link (src, decodebin);
 
@@ -185,7 +185,7 @@ stp_encode_from_msg (TranscodeServerCtx *ctx)
   /* The pads of decodebin and encodebin are dynamic,
    * so those will be linked when streams/pads are added */
 
-  /* When decodebin finds a stream that can be decoded, we check 
+  /* When decodebin finds a stream that can be decoded, we check
    * if we can pass that directly to encodebin instead of letting
    * decodebin find a decoding element automatically */
   g_signal_connect (decodebin, "autoplug-continue",
diff --git a/src/encode.h b/src/encode.h
index 79ed4aa..15ac859 100644
--- a/src/encode.h
+++ b/src/encode.h
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
diff --git a/src/lib.c b/src/lib.c
index 2d86b7c..4a3d4df 100644
--- a/src/lib.c
+++ b/src/lib.c
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
@@ -60,7 +60,7 @@ stp_on_gst_bus_message (GstBus             *bus,
 }
 
 /* When the incoming stream reaches EOS, we call this
- * which initiates a shutdown for all clients and then 
+ * which initiates a shutdown for all clients and then
  * the server itself */
 void
 stp_cleanup_transcode_server_ctx (TranscodeServerCtx *ctx)
@@ -107,8 +107,8 @@ pad_blocked_cleanup_cb (GstPad             *srcpad,
 }
 
 /* When a client leaves or is kicked, we set the response status
- * and then call cleanup here. This involves removing the appsink 
- * branch for the client from the pipeline, and then freeing the 
+ * and then call cleanup here. This involves removing the appsink
+ * branch for the client from the pipeline, and then freeing the
  * context. */
 void
 stp_cleanup_transcode_client_ctx (TranscodeClientCtx *ctx)
diff --git a/src/lib.h b/src/lib.h
index f1d9588..0fb3f02 100644
--- a/src/lib.h
+++ b/src/lib.h
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
@@ -39,7 +39,7 @@ struct _TranscodeServerCtx {
   SoupEncoding encoding;
   /* Set to TRUE when the incoming stream ends; let's us know if
    * we need to set the PUT response on EOS/ERROR on the pipeline,
-   * and whether to reject further data when using a persistent 
+   * and whether to reject further data when using a persistent
    * Content-Length + Content-Range PUT stream. */
   gboolean stream_finished;
   guint seconds_since_read;
diff --git a/src/main.c b/src/main.c
index 83e9609..e291b2c 100644
--- a/src/main.c
+++ b/src/main.c
@@ -1,4 +1,4 @@
-/* 
+/*
  * vim: set sts=2 sw=2 et :
  *
  * License: LGPL-2.1+
@@ -92,7 +92,7 @@ tee_src_pad_blocked_cb (GstPad             *srcpad,
   g_print ("Linked pads, removed probe. State was %s:%s.\n",
            gst_element_state_change_return_get_name (ret),
            gst_element_state_get_name (state));
- 
+
   if (gst_element_set_state (server_ctx->pipeline, GST_STATE_PLAYING) ==
       GST_STATE_CHANGE_FAILURE) {
     g_critical ("Unable to set pipeline back to PLAYING\n");
@@ -171,12 +171,12 @@ write_client_chunk_cb (GstElement         *appsink,
   GstBuffer *buffer;
   GstMemory *memory;
   gboolean eos;
-  
+
   if (!g_mutex_trylock (&ctx->can_write_chunk)) {
     g_print ("!");
     /* We cannot safely append to the message body till
      * the previous chunk is written out, otherwise we get
-     * a segfault. This is likely because SoupMessageBody 
+     * a segfault. This is likely because SoupMessageBody
      * uses a GSList for the chunks, which isn't MT-safe. */
     return GST_FLOW_OK;
   }
@@ -246,7 +246,7 @@ handle_request_cb (SoupServer        *server,
                    GHashTable        *ctx_table)
 {
   TranscodeServerCtx *server_ctx;
-  
+
   if (!msg)
     return;
 
@@ -282,7 +282,7 @@ GET:
     GstPad *srcpad, *sinkpadq2;
     TranscodeClientCtx *client_ctx;
     GstStateChangeReturn state;
- 
+
     server_ctx = get_server_ctx_from_msg (msg, ctx_table);
     if (server_ctx == NULL ||
         server_ctx->pipeline == NULL) {
@@ -306,13 +306,13 @@ GET:
       default:
         g_assert_not_reached ();
     }
- 
+
     /* Connect appsink to tee, and start streaming */
     client_ctx = g_new0 (TranscodeClientCtx, 1);
     client_ctx->msg = msg;
     client_ctx->server_ctx = server_ctx;
     g_mutex_init (&client_ctx->can_write_chunk);
- 
+
     bin = gst_bin_new (NULL);
 
     /* queue ! appsink */
@@ -344,7 +344,7 @@ GET:
     template = gst_pad_template_new ("teesrc", GST_PAD_SRC,
                                      GST_PAD_REQUEST, GST_CAPS_ANY);
     srcpad = gst_element_request_pad (tee, template, NULL, GST_CAPS_ANY);
-    
+
     gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK,
                        (GstPadProbeCallback) tee_src_pad_blocked_cb,
                        client_ctx, NULL);
@@ -366,8 +366,8 @@ GET:
 
     server_ctx->clients = g_list_prepend (server_ctx->clients, client_ctx);
 
-    /* Since we set the response as chunked, the server will 
-     * automatically pause the message for us, and the response 
+    /* Since we set the response as chunked, the server will
+     * automatically pause the message for us, and the response
      * won't be completed till we call soup_message_body_complete() */
     soup_message_set_status (msg, SOUP_STATUS_OK);
   }
@@ -421,13 +421,13 @@ got_first_request_body_chunk (SoupMessage        *msg,
   int num;
 
   /* Disconnect the ctx cleanup function */
-  num = g_signal_handlers_disconnect_by_func (msg, 
+  num = g_signal_handlers_disconnect_by_func (msg,
     G_CALLBACK (request_ended_no_body_cb), ctx);
   if (num != 1)
     g_critical ("Unable to remove signal handler for cleanup function!\n");
 
   /* Disconnect us from got-chunk */
-  num = g_signal_handlers_disconnect_by_func (msg, 
+  num = g_signal_handlers_disconnect_by_func (msg,
     G_CALLBACK (got_first_request_body_chunk), ctx);
   if (num != 1)
     g_critical ("Unable to remove signal handler for first request chunk!\n");
@@ -435,7 +435,7 @@ got_first_request_body_chunk (SoupMessage        *msg,
   ctx->msg = msg;
   ctx->pipeline = gst_pipeline_new ("pipe");
 
-  /* The chunked request is copied into this stream 
+  /* The chunked request is copied into this stream
    * for consumption by the gst pipeline */
 #ifdef PLAY_DEBUG
   stp_play_from_msg (ctx);
@@ -505,7 +505,7 @@ PUT: {
     connection_exists = g_hash_table_contains (ctx_table, soup_uri_get_path (uri));
 
     /* Data might be sent to us in one of three ways:
-     * 1. Chunked encoding with the stream (and request body) sent in chunks 
+     * 1. Chunked encoding with the stream (and request body) sent in chunks
      * 2. Content-Length encoding with the entire stream in one chunk
      * 3. A persistent Content-Length encoding connection, and subsequent
      *    requests can send more stream data, forever, subject to a timeout */
@@ -525,7 +525,7 @@ PUT: {
         g_print ("Content-Length encoding detected!\n");
         content_range = soup_message_headers_get_one (msg->request_headers,
                                                       "Content-Range");
-        /* TODO: Right now, we don't check if the Content-Range is valid, 
+        /* TODO: Right now, we don't check if the Content-Range is valid,
          * and just pass the data to the pipeline. */
         if (connection_exists && !content_range) {
           /* A connection already exists, and this one isn't a continuation
@@ -546,7 +546,7 @@ PUT: {
         g_assert_not_reached ();
     }
 
-    /* Whether the incoming stream is chunked or fixed length, we want to 
+    /* Whether the incoming stream is chunked or fixed length, we want to
      * handle it chunked, so tell libsoup to not collect chunks for
      * forming a complete request body */
     soup_message_body_set_accumulate (msg->request_body, FALSE);