Merge remote-tracking branch 'qatar/master'

* qatar/master:
  lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs
  lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs
  Add Dolby/DPLII downmix support to libavresample
  vorbisdec: replace div/mod in loop with a counter
  fate: vorbis: add 5.1 surround test
  rtpenc: Allow requesting H264 RTP packetization mode 0
  configure: Sort the library listings in the help text alphabetically
  dwt: remove variable-length arrays
  RTMPT protocol support
  http: Properly handle chunked transfer-encoding for replies to post data
  http: Fail reading if the connection has gone away
  amr: Mark an array const
  amr: More space cleanup
  rtpenc: Fix memory leaks in the muxer open function

Conflicts:
	Changelog
	configure
	doc/APIchanges
	libavformat/version.h

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

11 files changed, 303 insertions, 19 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 5e4f0022f2..3dca060994 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -373,6 +373,8 @@ OBJS-$(CONFIG_MMST_PROTOCOL)             += mmst.o mms.o asf.o
 OBJS-$(CONFIG_MD5_PROTOCOL)              += md5proto.o
 OBJS-$(CONFIG_PIPE_PROTOCOL)             += file.o
 OBJS-$(CONFIG_RTMP_PROTOCOL)             += rtmpproto.o rtmppkt.o
+OBJS-$(CONFIG_RTMPHTTP_PROTOCOL)         += rtmphttp.o
+OBJS-$(CONFIG_RTMPT_PROTOCOL)            += rtmpproto.o rtmppkt.o
 OBJS-$(CONFIG_RTP_PROTOCOL)              += rtpproto.o
 OBJS-$(CONFIG_SCTP_PROTOCOL)             += sctp.o
 OBJS-$(CONFIG_TCP_PROTOCOL)              += tcp.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 1862449a85..ed9227e5b6 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -277,6 +277,8 @@ void av_register_all(void)
     REGISTER_PROTOCOL (MD5,  md5);
     REGISTER_PROTOCOL (PIPE, pipe);
     REGISTER_PROTOCOL (RTMP, rtmp);
+    REGISTER_PROTOCOL (RTMPHTTP, rtmphttp);
+    REGISTER_PROTOCOL (RTMPT, rtmpt);
     REGISTER_PROTOCOL (RTP, rtp);
     REGISTER_PROTOCOL (SCTP, sctp);
     REGISTER_PROTOCOL (TCP, tcp);
diff --git a/libavformat/amr.c b/libavformat/amr.c
index 4ff6b8d13f..23272e4caa 100644
--- a/libavformat/amr.c
+++ b/libavformat/amr.c
@@ -124,8 +124,8 @@ static int amr_read_packet(AVFormatContext *s, AVPacket *pkt)
         };
 
         size = packed_size[mode] + 1;
-    } else if(enc->codec_id == CODEC_ID_AMR_WB) {
-        static uint8_t packed_size[16] = {
+    } else if (enc->codec_id == CODEC_ID_AMR_WB) {
+        static const uint8_t packed_size[16] = {
             18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1
         };
 
diff --git a/libavformat/http.c b/libavformat/http.c
index 9a5ca625b0..fb4a83ad0e 100644
--- a/libavformat/http.c
+++ b/libavformat/http.c
@@ -352,6 +352,8 @@ static int http_read_header(URLContext *h, int *new_location)
     char line[1024];
     int err = 0;
 
+    s->chunksize = -1;
+
     for (;;) {
         if ((err = http_get_line(s, line, sizeof(line))) < 0)
             return err;
@@ -470,7 +472,6 @@ static int http_connect(URLContext *h, const char *path, const char *local_path,
         s->http_code = 200;
         return 0;
     }
-    s->chunksize = -1;
 
     /* wait for header */
     err = http_read_header(h, new_location);
@@ -510,14 +511,13 @@ static int http_read(URLContext *h, uint8_t *buf, int size)
     HTTPContext *s = h->priv_data;
     int err, new_location;
 
-    if (s->end_chunked_post) {
-        if (!s->end_header) {
-            err = http_read_header(h, &new_location);
-            if (err < 0)
-                return err;
-        }
+    if (!s->hd)
+        return AVERROR_EOF;
 
-        return http_buf_read(h, buf, size);
+    if (s->end_chunked_post && !s->end_header) {
+        err = http_read_header(h, &new_location);
+        if (err < 0)
+            return err;
     }
 
     if (s->chunksize >= 0) {
diff --git a/libavformat/rtmphttp.c b/libavformat/rtmphttp.c
new file mode 100644
index 0000000000..c431d28853
--- /dev/null
+++ b/libavformat/rtmphttp.c
@@ -0,0 +1,239 @@
+/*
+ * RTMP HTTP network protocol
+ * Copyright (c) 2012 Samuel Pitoiset
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * RTMP HTTP protocol
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/intfloat.h"
+#include "libavutil/opt.h"
+#include "internal.h"
+#include "http.h"
+
+#define RTMPT_DEFAULT_PORT 80
+
+/* protocol handler context */
+typedef struct RTMP_HTTPContext {
+    URLContext   *stream;           ///< HTTP stream
+    char         host[256];         ///< hostname of the server
+    int          port;              ///< port to connect (default is 80)
+    char         client_id[64];     ///< client ID used for all requests except the first one
+    int          seq;               ///< sequence ID used for all requests
+    uint8_t      *out_data;         ///< output buffer
+    int          out_size;          ///< current output buffer size
+    int          out_capacity;      ///< current output buffer capacity
+    int          initialized;       ///< flag indicating when the http context is initialized
+    int          finishing;         ///< flag indicating when the client closes the connection
+} RTMP_HTTPContext;
+
+static int rtmp_http_send_cmd(URLContext *h, const char *cmd)
+{
+    RTMP_HTTPContext *rt = h->priv_data;
+    char uri[2048];
+    uint8_t c;
+    int ret;
+
+    ff_url_join(uri, sizeof(uri), "http", NULL, rt->host, rt->port,
+                "/%s/%s/%d", cmd, rt->client_id, rt->seq++);
+
+    av_opt_set_bin(rt->stream->priv_data, "post_data", rt->out_data,
+                   rt->out_size, 0);
+
+    /* send a new request to the server */
+    if ((ret = ff_http_do_new_request(rt->stream, uri)) < 0)
+        return ret;
+
+    /* re-init output buffer */
+    rt->out_size = 0;
+
+    /* read the first byte which contains the polling interval */
+    if ((ret = ffurl_read(rt->stream, &c, 1)) < 0)
+        return ret;
+
+    return ret;
+}
+
+static int rtmp_http_write(URLContext *h, const uint8_t *buf, int size)
+{
+    RTMP_HTTPContext *rt = h->priv_data;
+    void *ptr;
+
+    if (rt->out_size + size > rt->out_capacity) {
+        rt->out_capacity = (rt->out_size + size) * 2;
+        ptr = av_realloc(rt->out_data, rt->out_capacity);
+        if (!ptr)
+            return AVERROR(ENOMEM);
+        rt->out_data = ptr;
+    }
+
+    memcpy(rt->out_data + rt->out_size, buf, size);
+    rt->out_size += size;
+
+    return size;
+}
+
+static int rtmp_http_read(URLContext *h, uint8_t *buf, int size)
+{
+    RTMP_HTTPContext *rt = h->priv_data;
+    int ret, off = 0;
+
+    /* try to read at least 1 byte of data */
+    do {
+        ret = ffurl_read(rt->stream, buf + off, size);
+        if (ret < 0 && ret != AVERROR_EOF)
+            return ret;
+
+        if (ret == AVERROR_EOF) {
+            if (rt->finishing) {
+                /* Do not send new requests when the client wants to
+                 * close the connection. */
+                return AVERROR(EAGAIN);
+            }
+
+            /* When the client has reached end of file for the last request,
+             * we have to send a new request if we have buffered data.
+             * Otherwise, we have to send an idle POST. */
+            if (rt->out_size > 0) {
+                if ((ret = rtmp_http_send_cmd(h, "send")) < 0)
+                    return ret;
+            } else {
+                if ((ret = rtmp_http_write(h, "", 1)) < 0)
+                    return ret;
+
+                if ((ret = rtmp_http_send_cmd(h, "idle")) < 0)
+                    return ret;
+            }
+
+            if (h->flags & AVIO_FLAG_NONBLOCK) {
+                /* no incoming data to handle in nonblocking mode */
+                return AVERROR(EAGAIN);
+            }
+        } else {
+            off  += ret;
+            size -= ret;
+        }
+    } while (off <= 0);
+
+    return off;
+}
+
+static int rtmp_http_close(URLContext *h)
+{
+    RTMP_HTTPContext *rt = h->priv_data;
+    uint8_t tmp_buf[2048];
+    int ret = 0;
+
+    if (rt->initialized) {
+        /* client wants to close the connection */
+        rt->finishing = 1;
+
+        do {
+            ret = rtmp_http_read(h, tmp_buf, sizeof(tmp_buf));
+        } while (ret > 0);
+
+        /* re-init output buffer before sending the close command */
+        rt->out_size = 0;
+
+        if ((ret = rtmp_http_write(h, "", 1)) == 1)
+            ret = rtmp_http_send_cmd(h, "close");
+    }
+
+    av_freep(&rt->out_data);
+    ffurl_close(rt->stream);
+
+    return ret;
+}
+
+static int rtmp_http_open(URLContext *h, const char *uri, int flags)
+{
+    RTMP_HTTPContext *rt = h->priv_data;
+    char headers[1024], url[1024];
+    int ret, off = 0;
+
+    av_url_split(NULL, 0, NULL, 0, rt->host, sizeof(rt->host), &rt->port,
+                 NULL, 0, uri);
+
+    if (rt->port < 0)
+        rt->port = RTMPT_DEFAULT_PORT;
+
+    /* This is the first request that is sent to the server in order to
+     * register a client on the server and start a new session. The server
+     * replies with a unique id (usually a number) that is used by the client
+     * for all future requests.
+     * Note: the reply doesn't contain a value for the polling interval.
+     * A successful connect resets the consecutive index that is used
+     * in the URLs. */
+    ff_url_join(url, sizeof(url), "http", NULL, rt->host, rt->port, "/open/1");
+
+    /* alloc the http context */
+    if ((ret = ffurl_alloc(&rt->stream, url, AVIO_FLAG_READ_WRITE, NULL)) < 0)
+        goto fail;
+
+    /* set options */
+    snprintf(headers, sizeof(headers),
+             "Cache-Control: no-cache\r\n"
+             "Content-type: application/x-fcs\r\n"
+             "User-Agent: Shockwave Flash\r\n");
+    av_opt_set(rt->stream->priv_data, "headers", headers, 0);
+    av_opt_set(rt->stream->priv_data, "multiple_requests", "1", 0);
+    av_opt_set_bin(rt->stream->priv_data, "post_data", "", 1, 0);
+
+    /* open the http context */
+    if ((ret = ffurl_connect(rt->stream, NULL)) < 0)
+        goto fail;
+
+    /* read the server reply which contains a unique ID */
+    for (;;) {
+        ret = ffurl_read(rt->stream, rt->client_id + off, sizeof(rt->client_id) - off);
+        if (ret == AVERROR_EOF)
+            break;
+        if (ret < 0)
+            goto fail;
+        off += ret;
+        if (off == sizeof(rt->client_id)) {
+            ret = AVERROR(EIO);
+            goto fail;
+        }
+    }
+    while (off > 0 && isspace(rt->client_id[off - 1]))
+        off--;
+    rt->client_id[off] = '\0';
+
+    /* http context is now initialized */
+    rt->initialized = 1;
+    return 0;
+
+fail:
+    rtmp_http_close(h);
+    return ret;
+}
+
+URLProtocol ff_rtmphttp_protocol = {
+    .name           = "rtmphttp",
+    .url_open       = rtmp_http_open,
+    .url_read       = rtmp_http_read,
+    .url_write      = rtmp_http_write,
+    .url_close      = rtmp_http_close,
+    .priv_data_size = sizeof(RTMP_HTTPContext),
+    .flags          = URL_PROTOCOL_FLAG_NETWORK,
+};
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index 03f959e009..07ec62cc43 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -1112,9 +1112,15 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
     av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
                  path, sizeof(path), s->filename);
 
-    if (port < 0)
-        port = RTMP_DEFAULT_PORT;
-    ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
+    if (!strcmp(proto, "rtmpt")) {
+        /* open the http tunneling connection */
+        ff_url_join(buf, sizeof(buf), "rtmphttp", NULL, hostname, port, NULL);
+    } else {
+        /* open the tcp connection */
+        if (port < 0)
+            port = RTMP_DEFAULT_PORT;
+        ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
+    }
 
     if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
                           &s->interrupt_callback, NULL)) < 0) {
@@ -1425,3 +1431,21 @@ URLProtocol ff_rtmp_protocol = {
     .flags          = URL_PROTOCOL_FLAG_NETWORK,
     .priv_data_class= &rtmp_class,
 };
+
+static const AVClass rtmpt_class = {
+    .class_name = "rtmpt",
+    .item_name  = av_default_item_name,
+    .option     = rtmp_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+URLProtocol ff_rtmpt_protocol = {
+    .name            = "rtmpt",
+    .url_open        = rtmp_open,
+    .url_read        = rtmp_read,
+    .url_write       = rtmp_write,
+    .url_close       = rtmp_close,
+    .priv_data_size  = sizeof(RTMPContext),
+    .flags           = URL_PROTOCOL_FLAG_NETWORK,
+    .priv_data_class = &rtmpt_class,
+};
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 57d025a364..e16e610820 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -198,11 +198,11 @@ static int rtp_write_header(AVFormatContext *s1)
         /* max_header_toc_size + the largest AMR payload must fit */
         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
-            return -1;
+            goto fail;
         }
         if (st->codec->channels != 1) {
             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
-            return -1;
+            goto fail;
         }
     case CODEC_ID_AAC:
         s->num_frames = 0;
@@ -216,6 +216,10 @@ defaultcase:
     }
 
     return 0;
+
+fail:
+    av_freep(&s->buf);
+    return AVERROR(EINVAL);
 }
 
 /* send an rtcp sender report packet */
diff --git a/libavformat/rtpenc.h b/libavformat/rtpenc.h
index a83c633e73..8827cb3254 100644
--- a/libavformat/rtpenc.h
+++ b/libavformat/rtpenc.h
@@ -66,12 +66,14 @@ typedef struct RTPMuxContext RTPMuxContext;
 #define FF_RTP_FLAG_MP4A_LATM 1
 #define FF_RTP_FLAG_RFC2190   2
 #define FF_RTP_FLAG_SKIP_RTCP 4
+#define FF_RTP_FLAG_H264_MODE0 8
 
 #define FF_RTP_FLAG_OPTS(ctx, fieldname) \
     { "rtpflags", "RTP muxer flags", offsetof(ctx, fieldname), AV_OPT_TYPE_FLAGS, {.dbl = 0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
     { "latm", "Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_MP4A_LATM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
     { "rfc2190", "Use RFC 2190 packetization instead of RFC 4629 for H.263", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_RFC2190}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
     { "skip_rtcp", "Don't send RTCP sender reports", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_SKIP_RTCP}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
+    { "h264_mode0", "Use mode 0 for H264 in RTP", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_H264_MODE0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
 
 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
 
diff --git a/libavformat/rtpenc_h264.c b/libavformat/rtpenc_h264.c
index 86930bbac1..68f497590b 100644
--- a/libavformat/rtpenc_h264.c
+++ b/libavformat/rtpenc_h264.c
@@ -55,6 +55,12 @@ static void nal_send(AVFormatContext *s1, const uint8_t *buf, int size, int last
         uint8_t type = buf[0] & 0x1F;
         uint8_t nri = buf[0] & 0x60;
 
+        if (s->flags & FF_RTP_FLAG_H264_MODE0) {
+            av_log(s1, AV_LOG_ERROR,
+                   "NAL size %d > %d, try -slice-max-size %d\n", size,
+                   s->max_payload_size, s->max_payload_size);
+            return;
+        }
         av_log(s1, AV_LOG_DEBUG, "NAL size %d > %d\n", size, s->max_payload_size);
         s->buf[0] = 28;        /* FU Indicator; Type = 28 ---> FU-A */
         s->buf[0] |= nri;
diff --git a/libavformat/sdp.c b/libavformat/sdp.c
index caa661bbc2..5d5e9515f7 100644
--- a/libavformat/sdp.c
+++ b/libavformat/sdp.c
@@ -388,15 +388,20 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
     char *config = NULL;
 
     switch (c->codec_id) {
-        case CODEC_ID_H264:
+        case CODEC_ID_H264: {
+            int mode = 1;
+            if (fmt && fmt->oformat->priv_class &&
+                av_opt_flag_is_set(fmt->priv_data, "rtpflags", "h264_mode0"))
+                mode = 0;
             if (c->extradata_size) {
                 config = extradata2psets(c);
             }
             av_strlcatf(buff, size, "a=rtpmap:%d H264/90000\r\n"
-                                    "a=fmtp:%d packetization-mode=1%s\r\n",
+                                    "a=fmtp:%d packetization-mode=%d%s\r\n",
                                      payload_type,
-                                     payload_type, config ? config : "");
+                                     payload_type, mode, config ? config : "");
             break;
+        }
         case CODEC_ID_H263:
         case CODEC_ID_H263P:
             /* a=framesize is required by 3GPP TS 26.234 (PSS). It
diff --git a/libavformat/version.h b/libavformat/version.h
index cca6c1aea4..0b1bff2764 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFORMAT_VERSION_MAJOR 54
-#define LIBAVFORMAT_VERSION_MINOR  7
+#define LIBAVFORMAT_VERSION_MINOR  8
 #define LIBAVFORMAT_VERSION_MICRO 100
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \