[FFmpeg-soc] [soc]: r4782 - in rtmp: rtmppkt.c rtmpproto.c
kostya
subversion at mplayerhq.hu
Thu Jul 23 05:23:27 CEST 2009
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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