[FFmpeg-cvslog] RTMPT protocol support
Samuel Pitoiset
git at videolan.org
Mon Jun 18 20:25:39 CEST 2012
ffmpeg | branch: master | Samuel Pitoiset <samuel.pitoiset at gmail.com> | Sun Jun 17 20:24:43 2012 +0200| [8e50c57dcb481479f2fd46f9bdb6a9776b0d9fa6] | committer: Martin Storsjö
RTMPT protocol support
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8e50c57dcb481479f2fd46f9bdb6a9776b0d9fa6
---
Changelog | 1 +
configure | 4 +
doc/protocols.texi | 8 ++
libavformat/Makefile | 2 +
libavformat/allformats.c | 2 +
libavformat/rtmphttp.c | 239 ++++++++++++++++++++++++++++++++++++++++++++++
libavformat/rtmpproto.c | 30 +++++-
libavformat/version.h | 4 +-
8 files changed, 285 insertions(+), 5 deletions(-)
diff --git a/Changelog b/Changelog
index b80ff88..4288aa3 100644
--- a/Changelog
+++ b/Changelog
@@ -25,6 +25,7 @@ version <next>:
be used with -of old.
- Indeo Audio decoder
- channelsplit audio filter
+- RTMPT protocol support
version 0.8:
diff --git a/configure b/configure
index 4bb2030..d614366 100755
--- a/configure
+++ b/configure
@@ -1511,6 +1511,10 @@ mmsh_protocol_select="http_protocol"
mmst_protocol_deps="network"
rtmp_protocol_deps="!librtmp_protocol"
rtmp_protocol_select="tcp_protocol"
+rtmphttp_protocol_deps="!librtmp_protocol"
+rtmphttp_protocol_select="http_protocol"
+rtmpt_protocol_deps="!librtmp_protocol"
+rtmpt_protocol_select="rtmphttp_protocol"
rtp_protocol_select="udp_protocol"
sctp_protocol_deps="network netinet_sctp_h"
tcp_protocol_deps="network"
diff --git a/doc/protocols.texi b/doc/protocols.texi
index 8492033..0b4f1b1 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -243,6 +243,14 @@ For example to read with @command{avplay} a multimedia resource named
avplay rtmp://myserver/vod/sample
@end example
+ at section rtmpt
+
+Real-Time Messaging Protocol tunneled through HTTP.
+
+The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
+for streaming multimedia content within HTTP requests to traverse
+firewalls.
+
@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
diff --git a/libavformat/Makefile b/libavformat/Makefile
index ca4f7a0..6262324 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -345,6 +345,8 @@ 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 1320a28..42c588f 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -256,6 +256,8 @@ 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/rtmphttp.c b/libavformat/rtmphttp.c
new file mode 100644
index 0000000..fdcff50
--- /dev/null
+++ b/libavformat/rtmphttp.c
@@ -0,0 +1,239 @@
+/*
+ * RTMP HTTP network protocol
+ * Copyright (c) 2012 Samuel Pitoiset
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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 b3ae5a2..b3e2a30 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/version.h b/libavformat/version.h
index b00701e..aae0eb1 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -30,8 +30,8 @@
#include "libavutil/avutil.h"
#define LIBAVFORMAT_VERSION_MAJOR 54
-#define LIBAVFORMAT_VERSION_MINOR 3
-#define LIBAVFORMAT_VERSION_MICRO 1
+#define LIBAVFORMAT_VERSION_MINOR 4
+#define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \
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