Author: kostya
Date: Thu Jul 23 05:23:27 2009
New Revision: 4782

Log:
correct documentation and messages along with breaking long lines

Modified:
   rtmp/rtmppkt.c
   rtmp/rtmpproto.c

Modified: rtmp/rtmppkt.c
==============================================================================
--- rtmp/rtmppkt.c      Thu Jul 23 05:08:44 2009        (r4781)
+++ rtmp/rtmppkt.c      Thu Jul 23 05:23:27 2009        (r4782)
@@ -63,7 +63,9 @@ void ff_amf_write_field_name(uint8_t **d
 
 void ff_amf_write_object_end(uint8_t **dst)
 {
-    // first two bytes are field name length = 0, AMF object should end with 
it and end marker
+    /* first two bytes are field name length = 0,
+     * AMF object should end with it and end marker
+     */
     bytestream_put_be24(dst, AMF_DATA_TYPE_OBJECT_END);
 }
 

Modified: rtmp/rtmpproto.c
==============================================================================
--- rtmp/rtmpproto.c    Thu Jul 23 05:08:44 2009        (r4781)
+++ rtmp/rtmpproto.c    Thu Jul 23 05:23:27 2009        (r4782)
@@ -45,7 +45,7 @@
 
 /** RTMP protocol handler state */
 typedef enum {
-    STATE_START,      ///< client has not done anything
+    STATE_START,      ///< client has not done anything yet
     STATE_HANDSHAKED, ///< client has performed handshake
     STATE_CONNECTING, ///< client connected to server successfully
     STATE_READY,      ///< client has sent all needed commands and waits for 
server reply
@@ -56,10 +56,10 @@ typedef enum {
 typedef struct RTMPContext {
     URLContext*   stream;                     ///< TCP stream used in 
interactions with RTMP server
     RTMPPacket    prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when 
reading and sending packets
-    int           chunk_size;                 ///< size of chunks packet 
divided into
+    int           chunk_size;                 ///< size of the chunks RTMP 
packets are divided into
     char          playpath[256];              ///< path to filename to play 
(with possible "mp4:" prefix)
     ClientState   state;                      ///< current state
-    int           main_channel_id;            ///< an additional channel id 
which is used for some invokes
+    int           main_channel_id;            ///< an additional channel ID 
which is used for some invocations
     uint8_t*      flv_data;                   ///< buffer with data for demuxer
     int           flv_size;                   ///< current buffer size
     int           flv_off;                    ///< number of bytes read from 
current buffer
@@ -91,7 +91,7 @@ static const uint8_t rtmp_server_key[] =
 };
 
 /**
- * Generates 'connect' call and sends it to server.
+ * Generates 'connect' call and sends it to the server.
  */
 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
                         const char *host, int port, const char *app)
@@ -134,8 +134,8 @@ static void gen_connect(URLContext *s, R
 }
 
 /**
- * Generates 'createStream' call and sends it to server. It should make server
- * allocate some channel for media streams.
+ * Generates 'createStream' call and sends it to the server. It should make
+ * the server allocate some channel for media streams.
  */
 static void gen_create_stream(URLContext *s, RTMPContext *rt)
 {
@@ -155,7 +155,7 @@ static void gen_create_stream(URLContext
 }
 
 /**
- * Generates 'play' call and sends it to server, then pings server
+ * Generates 'play' call and sends it to the server, then pings the server
  * to start actual playing.
  */
 static void gen_play(URLContext *s, RTMPContext *rt)
@@ -191,7 +191,7 @@ static void gen_play(URLContext *s, RTMP
 }
 
 /**
- * Generates ping reply and sends it to server.
+ * Generates ping reply and sends it to the server.
  */
 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
 {
@@ -297,7 +297,8 @@ static int rtmp_validate_digest(uint8_t 
 }
 
 /**
- * Performs handshake with server by means of exchanging HMAC-SHA2 signed 
ppseudorangom data.
+ * Performs handshake with the server by means of exchanging pseudorandom data
+ * signed with HMAC-SHA2 digest.
  */
 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
 {
@@ -375,8 +376,8 @@ static int rtmp_handshake(URLContext *s,
 
 /**
  * Parses received packet and may perform some action depending on packet 
contents.
- * @return 0 for no errors, -1 for serious errors which prevent further 
communications,
- *         positive values for not critical errors
+ * @return 0 for no errors, -1 for serious errors which prevent further
+ *         communications, positive values for uncritical errors
  */
 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
 {
@@ -385,8 +386,8 @@ static int rtmp_parse_result(URLContext 
     switch (pkt->type) {
     case RTMP_PT_CHUNK_SIZE:
         if (pkt->data_size != 4) {
-            av_log(LOG_CONTEXT, AV_LOG_ERROR, "Chunk size change packet is not 
4 (%d)\n",
-                   pkt->data_size);
+            av_log(LOG_CONTEXT, AV_LOG_ERROR,
+                   "Chunk size change packet is not 4 bytes long (%d)\n", 
pkt->data_size);
             return -1;
         }
         rt->chunk_size = AV_RB32(pkt->data);
@@ -529,11 +530,13 @@ static int get_packet(URLContext *s, int
 }
 
 /**
- * Opens RTMP connection and verifies that stream can be played.
+ * Opens RTMP connection and verifies that the stream can be played.
  *
  * URL syntax: rtmp://server[:port][/app][/playpath]
- *             where 'app' is first one or two directories in the path 
(/ondemand/, /flash/live/, etc)
- *             and 'playpath' is file name (the rest of path, may be prefixed 
with "mp4:")
+ *             where 'app' is first one or two directories in the path
+ *             (e.g. /ondemand/, /flash/live/, etc.)
+ *             and 'playpath' is a file name (the rest of the path,
+ *             may be prefixed with "mp4:")
  */
 static int rtmp_open(URLContext *s, const char *uri, int flags)
 {
@@ -560,7 +563,7 @@ static int rtmp_open(URLContext *s, cons
         goto fail;
 
     if (!is_input) {
-        av_log(LOG_CONTEXT, AV_LOG_ERROR, "RTMP output is not supported 
yet\n");
+        av_log(LOG_CONTEXT, AV_LOG_ERROR, "RTMP output is not supported 
yet.\n");
         goto fail;
     } else {
         rt->state = STATE_START;
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