Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
  rtmp: Move the CONFIG_ condition into the if conditions
  aac: Mention abbreviation as well in long_name
  build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
  doc: Add Git configuration section
  configure: Add a dependency on https for rtmpts
  rtp: Only choose static payload types if the sample rate and channels are right

Conflicts:
	doc/git-howto.texi
	libavformat/rtmpproto.c

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

5 files changed, 23 insertions, 12 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 6a40fad3ef..985da20ef0 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -398,6 +398,7 @@ OBJS-$(CONFIG_TCP_PROTOCOL)              += tcp.o
 OBJS-$(CONFIG_TLS_PROTOCOL)              += tls.o
 OBJS-$(CONFIG_UDP_PROTOCOL)              += udp.o
 
+SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
 SKIPHEADERS-$(CONFIG_NETWORK)            += network.h rtsp.h
 TESTPROGS = seek
 
diff --git a/libavformat/rtmpcrypt.c b/libavformat/rtmpcrypt.c
index 7dee0ad2a0..06574164fe 100644
--- a/libavformat/rtmpcrypt.c
+++ b/libavformat/rtmpcrypt.c
@@ -1,5 +1,7 @@
 /*
  * RTMPE network protocol
+ * Copyright (c) 2008-2009 Andrej Stepanchuk
+ * Copyright (c) 2009-2010 Howard Chu
  * Copyright (c) 2012 Samuel Pitoiset
  *
  * This file is part of FFmpeg.
diff --git a/libavformat/rtmpdh.c b/libavformat/rtmpdh.c
index fad87fca3b..03582eafb9 100644
--- a/libavformat/rtmpdh.c
+++ b/libavformat/rtmpdh.c
@@ -1,5 +1,7 @@
 /*
  * RTMP Diffie-Hellmann utilities
+ * Copyright (c) 2009 Andrej Stepanchuk
+ * Copyright (c) 2009-2010 Howard Chu
  * Copyright (c) 2012 Samuel Pitoiset
  *
  * This file is part of FFmpeg.
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index a85ed31a5f..4ffd48eeec 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -719,7 +719,6 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
     int i;
     int server_pos, client_pos;
     uint8_t digest[32], signature[32];
-#define encrypted (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted)
     int ret, type = 0;
 
     av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
@@ -729,7 +728,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
     for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
         tosend[i] = av_lfg_get(&rnd) >> 24;
 
-    if (encrypted) {
+    if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
         /* When the client wants to use RTMPE, we have to change the command
          * byte to 0x06 which means to use encrypted data and we have to set
          * the flash version to at least 9.0.115.0. */
@@ -745,7 +744,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
             return ret;
     }
 
-    client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, encrypted);
+    client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
     if (client_pos < 0)
         return client_pos;
 
@@ -799,7 +798,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
         if (ret < 0)
             return ret;
 
-        if (encrypted) {
+        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
             /* Compute the shared secret key sent by the server and initialize
              * the RC4 encryption. */
             if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
@@ -829,7 +828,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
         if (ret < 0)
             return ret;
 
-        if (encrypted) {
+        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
             /* Encrypt the signature to be send to the server. */
             ff_rtmpe_encrypt_sig(rt->stream, tosend +
                                  RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
@@ -841,13 +840,13 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
                                RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
             return ret;
 
-        if (encrypted) {
+        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
             /* Set RC4 keys for encryption and update the keystreams. */
             if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
                 return ret;
         }
     } else {
-        if (encrypted) {
+        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
             /* Compute the shared secret key sent by the server and initialize
              * the RC4 encryption. */
             if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
@@ -865,7 +864,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
                                RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
             return ret;
 
-        if (encrypted) {
+        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
             /* Set RC4 keys for encryption and update the keystreams. */
             if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
                 return ret;
@@ -874,7 +873,6 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
 
     return 0;
 }
-#undef encrypted
 
 /**
  * Parse received packet and possibly perform some action depending on
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index a5484ae6fa..2a801625e9 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -110,9 +110,17 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec)
                 !fmt->oformat->priv_class ||
                 !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
                 continue;
-            if (codec->codec_id == CODEC_ID_PCM_S16BE)
-                if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
-                    continue;
+            /* G722 has 8000 as nominal rate even if the sample rate is 16000,
+             * see section 4.5.2 in RFC 3551. */
+            if (codec->codec_id == CODEC_ID_ADPCM_G722 &&
+                codec->sample_rate == 16000 && codec->channels == 1)
+                return AVRtpPayloadTypes[i].pt;
+            if (codec->codec_type == AVMEDIA_TYPE_AUDIO &&
+                ((AVRtpPayloadTypes[i].clock_rate > 0 &&
+                  codec->sample_rate != AVRtpPayloadTypes[i].clock_rate) ||
+                 (AVRtpPayloadTypes[i].audio_channels > 0 &&
+                  codec->channels != AVRtpPayloadTypes[i].audio_channels)))
+                continue;
             return AVRtpPayloadTypes[i].pt;
         }