Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtmp: Add a new option 'rtmp_pageurl'
  doc: Update the description of the rtmp_tcurl option
  rtmp: Make the description of the rtmp_tcurl option more generic
  libfdk-aacenc: add LATM/LOAS encapsulation support
  sctp: add port missing error message
  tcp: add port missing error message
  avfilter: Fix printf format string conversion specifier

Conflicts:
	libavcodec/version.h
	libavfilter/avfilter.c

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

doc/protocols.texi

@@ -259,6 +259,10 @@ playpath. If a live stream of that name is not found, it plays the
 recorded stream. The other possible values are @code{live} and
 @code{recorded}.
 
+@item rtmp_pageurl
+URL of the web page in which the media was embedded. By default no
+value will be sent.
+
 @item rtmp_playpath
 Stream identifier to play or to publish. This option overrides the
 parameter specified in the URI.
@@ -267,7 +271,7 @@ parameter specified in the URI.
 URL of the SWF player for the media. By default no value will be sent.
 
 @item rtmp_tcurl
-URL of the target stream.
+URL of the target stream. Defaults to proto://host[:port]/app.
 
 @end table
 

libavcodec/libfdk-aacenc.c

@@ -33,6 +33,8 @@ typedef struct AACContext {
     int afterburner;
     int eld_sbr;
     int signaling;
+    int latm;
+    int header_period;
 
     AudioFrameQueue afq;
 } AACContext;
@@ -45,6 +47,8 @@ static const AVOption aac_enc_options[] = {
     { "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
     { "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
     { "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
+    { "latm", "Output LATM/LOAS encapsulated data", offsetof(AACContext, latm), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
+    { "header_period", "StreamMuxConfig and PCE repetition period (in frames)", offsetof(AACContext, header_period), AV_OPT_TYPE_INT, { 0 }, 0, 0xffff, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
     { NULL }
 };
 
@@ -204,12 +208,21 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     /* Choose bitstream format - if global header is requested, use
      * raw access units, otherwise use ADTS. */
     if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
-                                   avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : 2)) != AACENC_OK) {
+                                   avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) {
         av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
                aac_get_error(err));
         goto error;
     }
 
+    if (s->latm && s->header_period) {
+        if ((err = aacEncoder_SetParam(s->handle, AACENC_HEADER_PERIOD,
+                                       s->header_period)) != AACENC_OK) {
+             av_log(avctx, AV_LOG_ERROR, "Unable to set header period: %s\n",
+                    aac_get_error(err));
+             goto error;
+        }
+    }
+
     /* If no signaling mode is chosen, use explicit hierarchical signaling
      * if using mp4 mode (raw access units, with global header) and
      * implicit signaling if using ADTS. */

libavcodec/version.h

@@ -28,7 +28,7 @@
 
 #define LIBAVCODEC_VERSION_MAJOR 54
 #define LIBAVCODEC_VERSION_MINOR  44
-#define LIBAVCODEC_VERSION_MICRO 100
+#define LIBAVCODEC_VERSION_MICRO 101
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \

libavformat/rtmpproto.c

@@ -90,6 +90,7 @@ typedef struct RTMPContext {
     char*         tcurl;                      ///< url of the target stream
     char*         flashver;                   ///< version of the flash plugin
     char*         swfurl;                     ///< url of the swf player
+    char*         pageurl;                    ///< url of the web page
     int           server_bw;                  ///< server bandwidth
     int           client_buffer_time;         ///< client buffer time in ms
     int           flush_interval;             ///< number of packets flushed in the same request (RTMPT only)
@@ -232,6 +233,11 @@ static int gen_connect(URLContext *s, RTMPContext *rt)
         ff_amf_write_number(&p, 252.0);
         ff_amf_write_field_name(&p, "videoFunction");
         ff_amf_write_number(&p, 1.0);
+
+        if (rt->pageurl) {
+            ff_amf_write_field_name(&p, "pageUrl");
+            ff_amf_write_string(&p, rt->pageurl);
+        }
     }
     ff_amf_write_object_end(&p);
 
@@ -1498,9 +1504,10 @@ static const AVOption rtmp_options[] = {
     {"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_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
     {"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},
+    {"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
     { NULL },
 };
 

libavformat/sctp.c

@@ -170,8 +170,12 @@ static int sctp_open(URLContext *h, const char *uri, int flags)
 
     av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname),
                  &port, path, sizeof(path), uri);
-    if (strcmp(proto,"sctp") || port <= 0 || port >= 65536)
+    if (strcmp(proto, "sctp"))
         return AVERROR(EINVAL);
+    if (port <= 0 || port >= 65536) {
+        av_log(s, AV_LOG_ERROR, "Port missing in uri\n");
+        return AVERROR(EINVAL);
+    }
 
     s->max_streams = 0;
     p = strchr(uri, '?');

libavformat/tcp.c

@@ -49,9 +49,12 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
 
     av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname),
         &port, path, sizeof(path), uri);
-    if (strcmp(proto,"tcp") || port <= 0 || port >= 65536)
+    if (strcmp(proto, "tcp"))
         return AVERROR(EINVAL);
-
+    if (port <= 0 || port >= 65536) {
+        av_log(h, AV_LOG_ERROR, "Port missing in uri\n");
+        return AVERROR(EINVAL);
+    }
     p = strchr(uri, '?');
     if (p) {
         if (av_find_info_tag(buf, sizeof(buf), "listen", p))