Merge remote-tracking branch 'qatar/master'

* qatar/master: (22 commits)
  configure: add check for w32threads to enable it automatically
  rtmp: do not hardcode invoke numbers
  cinepack: return non-generic errors
  fate-lavf-ts: use -mpegts_transport_stream_id option.
  Add an APIchanges entry and a minor bump for avio changes.
  avio: Mark the old interrupt callback mechanism as deprecated
  avplay: Set the new interrupt callback
  avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
  cinepak: remove redundant coordinate checks
  cinepak: check strip_size
  cinepak, simplify, use AV_RB24()
  cinepak: simplify, use FFMIN()
  cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
  applehttp: Fix seeking in streams not starting at DTS=0
  http: Don't use the normal http proxy mechanism for https
  tls: Handle connection via a http proxy
  http: Reorder two code blocks
  http: Add a new protocol for opening connections via http proxies
  http: Split out the non-chunked buffer reading part from http_read
  segafilm: add support for raw videos
  ...

Conflicts:
	avconv.c
	configure
	doc/APIchanges
	libavcodec/cinepak.c
	libavformat/applehttp.c
	libavformat/version.h
	tests/lavf-regression.sh
	tests/ref/lavf/ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 5 insertions, 4 deletions

diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index e73158f12c..8935f56369 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -74,6 +74,7 @@ typedef struct RTMPContext {
     int           skip_bytes;                 ///< number of bytes to skip from the input FLV stream in the next write call
     uint8_t       flv_header[11];             ///< partial incoming flv packet header
     int           flv_header_bytes;           ///< number of initialized bytes in flv_header
+    int           nb_invokes;                 ///< keeps track of invoke messages
 } RTMPContext;
 
 #define PLAYER_KEY_OPEN_PART_LEN 30   ///< length of partial key used for first client digest signing
@@ -166,7 +167,7 @@ static void gen_release_stream(URLContext *s, RTMPContext *rt)
     av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
     p = pkt.data;
     ff_amf_write_string(&p, "releaseStream");
-    ff_amf_write_number(&p, 2.0);
+    ff_amf_write_number(&p, ++rt->nb_invokes);
     ff_amf_write_null(&p);
     ff_amf_write_string(&p, rt->playpath);
 
@@ -189,7 +190,7 @@ static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
     av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
     p = pkt.data;
     ff_amf_write_string(&p, "FCPublish");
-    ff_amf_write_number(&p, 3.0);
+    ff_amf_write_number(&p, ++rt->nb_invokes);
     ff_amf_write_null(&p);
     ff_amf_write_string(&p, rt->playpath);
 
@@ -212,7 +213,7 @@ static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
     av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
     p = pkt.data;
     ff_amf_write_string(&p, "FCUnpublish");
-    ff_amf_write_number(&p, 5.0);
+    ff_amf_write_number(&p, ++rt->nb_invokes);
     ff_amf_write_null(&p);
     ff_amf_write_string(&p, rt->playpath);
 
@@ -234,7 +235,7 @@ static void gen_create_stream(URLContext *s, RTMPContext *rt)
 
     p = pkt.data;
     ff_amf_write_string(&p, "createStream");
-    ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
+    ff_amf_write_number(&p, ++rt->nb_invokes);
     ff_amf_write_null(&p);
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);