Module: libav
Branch: master
Commit: d4aef997809167832ecc64e89dda8cb445e5fe10

Author:    Josh Allmann <joshua.allm...@gmail.com>
Committer: Martin Storsjö <mar...@martin.st>
Date:      Mon Sep 16 13:20:57 2013 -0700

rtmp: Follow Flash player numbering for channels.

Channel 4 is typically used by the Flash player to transmit
audio, channel 6 for video, and various stream-specific invokes
get sent over channel 8, which is designated the source channel.

This more closely matches the behavior of the Flash player,
including the transmission of play requests over channel 8.

Signed-off-by: Martin Storsjö <mar...@martin.st>

---

 libavformat/rtmppkt.h   |    4 ++--
 libavformat/rtmpproto.c |    4 ++--
 2 files changed, 4 insertions(+), 4 deletions(-)

diff --git a/libavformat/rtmppkt.h b/libavformat/rtmppkt.h
index ff5d171..e3120be 100644
--- a/libavformat/rtmppkt.h
+++ b/libavformat/rtmppkt.h
@@ -36,9 +36,9 @@
 enum RTMPChannel {
     RTMP_NETWORK_CHANNEL = 2,   ///< channel for network-related messages 
(bandwidth report, ping, etc)
     RTMP_SYSTEM_CHANNEL,        ///< channel for sending server control 
messages
-    RTMP_SOURCE_CHANNEL,        ///< channel for sending a/v to server
-    RTMP_VIDEO_CHANNEL = 8,     ///< channel for video data
     RTMP_AUDIO_CHANNEL,         ///< channel for audio data
+    RTMP_VIDEO_CHANNEL   = 6,   ///< channel for video data
+    RTMP_SOURCE_CHANNEL  = 8,   ///< channel for a/v invokes
 };
 
 /**
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index f9412bf..bda4798 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -693,7 +693,7 @@ static int gen_play(URLContext *s, RTMPContext *rt)
 
     av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
 
-    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE,
+    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
                                      0, 29 + strlen(rt->playpath))) < 0)
         return ret;
 
@@ -2637,7 +2637,7 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, 
int size)
         if (rt->flv_header_bytes < 11) {
             const uint8_t *header = rt->flv_header;
             int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
-            int channel = RTMP_SOURCE_CHANNEL;
+            int channel = RTMP_AUDIO_CHANNEL;
             bytestream_get_buffer(&buf_temp, rt->flv_header + 
rt->flv_header_bytes, copy);
             rt->flv_header_bytes += copy;
             size_temp            -= copy;

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