Merge remote-tracking branch 'qatar/master'

* qatar/master: (25 commits)
  rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC
  ape: Use unsigned integer maths
  arm: dsputil: fix overreads in put/avg_pixels functions
  h264: K&R formatting cosmetics for header files (part II/II)
  h264: K&R formatting cosmetics for header files (part I/II)
  rtmp: Implement check bandwidth notification.
  rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player.
  rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin.
  rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream.
  cmdutils: Add fallback case to switch in check_stream_specifier().
  sctp: be consistent with socket option level
  configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags.
  vcr1enc: drop pointless empty encode_init() wrapper function
  vcr1: drop pointless write-only AVCodecContext member from VCR1Context
  vcr1: group encoder code together to save #ifdefs
  vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments
  mov: make one comment slightly more specific
  lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX
  lavfi: move audio-related functions to a separate file.
  lavfi: remove some audio-related function from public API.
  ...

Conflicts:
	cmdutils.c
	libavcodec/h264.h
	libavcodec/h264_mvpred.h
	libavcodec/vcr1.c
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/defaults.c
	libavfilter/internal.h

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

3 files changed, 61 insertions, 15 deletions

diff --git a/libavformat/mov.c b/libavformat/mov.c
index 240bccf402..9a2a59f36c 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -1051,7 +1051,7 @@ static int mov_read_glbl(MOVContext *c, AVIOContext *pb, MOVAtom atom)
         return AVERROR_INVALIDDATA;
 
     if (atom.size >= 10) {
-        // Broken files created by legacy versions of Libav and FFmpeg will
+        // Broken files created by legacy versions of libavformat will
         // wrap a whole fiel atom inside of a glbl atom.
         unsigned size = avio_rb32(pb);
         unsigned type = avio_rl32(pb);
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index 2feb2409ba..c7db91de9e 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -44,6 +44,8 @@
 
 #define APP_MAX_LENGTH 128
 #define PLAYPATH_MAX_LENGTH 256
+#define TCURL_MAX_LENGTH 512
+#define FLASHVER_MAX_LENGTH 64
 
 /** RTMP protocol handler state */
 typedef enum {
@@ -82,6 +84,9 @@ typedef struct RTMPContext {
     int           flv_header_bytes;           ///< number of initialized bytes in flv_header
     int           nb_invokes;                 ///< keeps track of invoke messages
     int           create_stream_invoke;       ///< invoke id for the create stream command
+    char*         tcurl;                      ///< url of the target stream
+    char*         flashver;                   ///< version of the flash plugin
+    char*         swfurl;                     ///< url of the swf player
 } RTMPContext;
 
 #define PLAYER_KEY_OPEN_PART_LEN 30   ///< length of partial key used for first client digest signing
@@ -110,35 +115,34 @@ static const uint8_t rtmp_server_key[] = {
 /**
  * Generate 'connect' call and send it to the server.
  */
-static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
-                        const char *host, int port)
+static void gen_connect(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
-    uint8_t ver[64], *p;
-    char tcurl[512];
+    uint8_t *p;
 
     ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
     p = pkt.data;
 
-    ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
     ff_amf_write_string(&p, "connect");
     ff_amf_write_number(&p, ++rt->nb_invokes);
     ff_amf_write_object_start(&p);
     ff_amf_write_field_name(&p, "app");
     ff_amf_write_string(&p, rt->app);
 
-    if (rt->is_input) {
-        snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
-                 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
-    } else {
-        snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
+    if (!rt->is_input) {
         ff_amf_write_field_name(&p, "type");
         ff_amf_write_string(&p, "nonprivate");
     }
     ff_amf_write_field_name(&p, "flashVer");
-    ff_amf_write_string(&p, ver);
+    ff_amf_write_string(&p, rt->flashver);
+
+    if (rt->swfurl) {
+        ff_amf_write_field_name(&p, "swfUrl");
+        ff_amf_write_string(&p, rt->swfurl);
+    }
+
     ff_amf_write_field_name(&p, "tcUrl");
-    ff_amf_write_string(&p, tcurl);
+    ff_amf_write_string(&p, rt->tcurl);
     if (rt->is_input) {
         ff_amf_write_field_name(&p, "fpad");
         ff_amf_write_bool(&p, 0);
@@ -369,6 +373,25 @@ static void gen_server_bw(URLContext *s, RTMPContext *rt)
 }
 
 /**
+ * Generate check bandwidth message and send it to the server.
+ */
+static void gen_check_bw(URLContext *s, RTMPContext *rt)
+{
+    RTMPPacket pkt;
+    uint8_t *p;
+
+    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 21);
+
+    p = pkt.data;
+    ff_amf_write_string(&p, "_checkbw");
+    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]);
+    ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
  * Generate report on bytes read so far and send it to the server.
  */
 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
@@ -687,6 +710,8 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
             if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
             if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
             if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
+        } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
+            gen_check_bw(s, rt);
         }
         break;
     }
@@ -910,13 +935,31 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
         strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
     }
 
+    if (!rt->tcurl) {
+        rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
+        ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
+                    port, "/%s", rt->app);
+    }
+
+    if (!rt->flashver) {
+        rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
+        if (rt->is_input) {
+            snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
+                    RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
+                    RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
+        } else {
+            snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
+                    "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
+        }
+    }
+
     rt->client_report_size = 1048576;
     rt->bytes_read = 0;
     rt->last_bytes_read = 0;
 
     av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
            proto, path, rt->app, rt->playpath);
-    gen_connect(s, rt, proto, hostname, port);
+    gen_connect(s, rt);
 
     do {
         ret = get_packet(s, 1);
@@ -1052,11 +1095,14 @@ static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
 
 static const AVOption rtmp_options[] = {
     {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+    {"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
     {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
     {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
     {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
     {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+    {"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+    {"rtmp_tcurl", "URL of the target stream. Defaults to rtmp://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     { NULL },
 };
 
diff --git a/libavformat/sctp.c b/libavformat/sctp.c
index 3823e03ebe..817b0049a9 100644
--- a/libavformat/sctp.c
+++ b/libavformat/sctp.c
@@ -227,7 +227,7 @@ static int sctp_open(URLContext *h, const char *uri, int flags)
     if (s->max_streams) {
         initparams.sinit_max_instreams = s->max_streams;
         initparams.sinit_num_ostreams  = s->max_streams;
-        if (setsockopt(fd, SOL_SCTP, SCTP_INITMSG, &initparams,
+        if (setsockopt(fd, IPPROTO_SCTP, SCTP_INITMSG, &initparams,
                        sizeof(initparams)) < 0)
             av_log(h, AV_LOG_ERROR,
                    "SCTP ERROR: Unable to initialize socket max streams %d\n",