From d4aef997809167832ecc64e89dda8cb445e5fe10 Mon Sep 17 00:00:00 2001
From: Josh Allmann <joshua.allmann@gmail.com>
Date: Mon, 16 Sep 2013 13:20:57 -0700
Subject: [PATCH] rtmp: Follow Flash player numbering for channels.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

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 <martin@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 ff5d1716dfe..e3120becc93 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 f9412bf2c2e..bda4798b69c 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;
-- 
GitLab