[FFmpeg-devel] [ONLY FOR TEST PATCH 2/2] avformat/rtmp: add rtmp over srt
Zhao Zhili
quinkblack at foxmail.com
Tue May 18 19:03:21 EEST 2021
---
Test with:
./ffplay -listen 1 rtmpsrt://127.0.0.1:8888
./ffmpeg -re -i bunny.mp4 -c copy -f flv rtmpsrt://127.0.0.1:8888
configure | 2 +
libavformat/Makefile | 2 +
libavformat/protocols.c | 2 +
libavformat/rtmpproto.c | 11 ++-
libavformat/rtmpsrt.c | 167 ++++++++++++++++++++++++++++++++++++++++
5 files changed, 183 insertions(+), 1 deletion(-)
create mode 100644 libavformat/rtmpsrt.c
diff --git a/configure b/configure
index 82367fd30d..76cd56477a 100755
--- a/configure
+++ b/configure
@@ -3476,6 +3476,7 @@ ffrtmpcrypt_protocol_deps_any="gcrypt gmp openssl mbedtls"
ffrtmpcrypt_protocol_select="tcp_protocol"
ffrtmphttp_protocol_conflict="librtmp_protocol"
ffrtmphttp_protocol_select="http_protocol"
+ffrtmpsrt_protocol_select="libsrt_protocol"
ftp_protocol_select="tcp_protocol"
gopher_protocol_select="tcp_protocol"
gophers_protocol_select="tls_protocol"
@@ -3502,6 +3503,7 @@ rtmpte_protocol_select="ffrtmpcrypt_protocol ffrtmphttp_protocol"
rtmpte_protocol_suggest="zlib"
rtmpts_protocol_select="ffrtmphttp_protocol https_protocol"
rtmpts_protocol_suggest="zlib"
+rtmpsrt_protocol_select="ffrtmpsrt_protocol"
rtp_protocol_select="udp_protocol"
schannel_conflict="openssl gnutls libtls mbedtls"
sctp_protocol_deps="struct_sctp_event_subscribe struct_msghdr_msg_flags"
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 85b5d8e7eb..7770fb2f8c 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -618,6 +618,7 @@ OBJS-$(CONFIG_CRYPTO_PROTOCOL) += crypto.o
OBJS-$(CONFIG_DATA_PROTOCOL) += data_uri.o
OBJS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpcrypt.o rtmpdigest.o rtmpdh.o
OBJS-$(CONFIG_FFRTMPHTTP_PROTOCOL) += rtmphttp.o
+OBJS-$(CONFIG_FFRTMPSRT_PROTOCOL) += rtmpsrt.o
OBJS-$(CONFIG_FILE_PROTOCOL) += file.o
OBJS-$(CONFIG_FTP_PROTOCOL) += ftp.o urldecode.o
OBJS-$(CONFIG_GOPHER_PROTOCOL) += gopher.o
@@ -638,6 +639,7 @@ OBJS-$(CONFIG_RTMPS_PROTOCOL) += rtmpproto.o rtmpdigest.o rtmppkt.o
OBJS-$(CONFIG_RTMPT_PROTOCOL) += rtmpproto.o rtmpdigest.o rtmppkt.o
OBJS-$(CONFIG_RTMPTE_PROTOCOL) += rtmpproto.o rtmpdigest.o rtmppkt.o
OBJS-$(CONFIG_RTMPTS_PROTOCOL) += rtmpproto.o rtmpdigest.o rtmppkt.o
+OBJS-$(CONFIG_RTMPSRT_PROTOCOL) += rtmpproto.o rtmpdigest.o rtmppkt.o
OBJS-$(CONFIG_RTP_PROTOCOL) += rtpproto.o ip.o
OBJS-$(CONFIG_SCTP_PROTOCOL) += sctp.o
OBJS-$(CONFIG_SRTP_PROTOCOL) += srtpproto.o srtp.o
diff --git a/libavformat/protocols.c b/libavformat/protocols.c
index 4b6b1c8e98..3f848338b0 100644
--- a/libavformat/protocols.c
+++ b/libavformat/protocols.c
@@ -31,6 +31,7 @@ extern const URLProtocol ff_crypto_protocol;
extern const URLProtocol ff_data_protocol;
extern const URLProtocol ff_ffrtmpcrypt_protocol;
extern const URLProtocol ff_ffrtmphttp_protocol;
+extern const URLProtocol ff_ffrtmpsrt_protocol;
extern const URLProtocol ff_file_protocol;
extern const URLProtocol ff_ftp_protocol;
extern const URLProtocol ff_gopher_protocol;
@@ -51,6 +52,7 @@ extern const URLProtocol ff_rtmps_protocol;
extern const URLProtocol ff_rtmpt_protocol;
extern const URLProtocol ff_rtmpte_protocol;
extern const URLProtocol ff_rtmpts_protocol;
+extern const URLProtocol ff_rtmpsrt_protocol;
extern const URLProtocol ff_rtp_protocol;
extern const URLProtocol ff_sctp_protocol;
extern const URLProtocol ff_srtp_protocol;
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index 5a540e3240..50e41662e8 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -128,6 +128,7 @@ typedef struct RTMPContext {
char auth_params[500];
int do_reconnect;
int auth_tried;
+ int rtmp_over_srt;
} RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
@@ -2624,7 +2625,7 @@ static int rtmp_open(URLContext *s, const char *uri, int flags, AVDictionary **o
}
}
- if (rt->listen && strcmp(proto, "rtmp")) {
+ if (rt->listen && strcmp(proto, "rtmp") && strcmp(proto, "rtmpsrt")) {
av_log(s, AV_LOG_ERROR, "rtmp_listen not available for %s\n",
proto);
return AVERROR(EINVAL);
@@ -2647,6 +2648,12 @@ static int rtmp_open(URLContext *s, const char *uri, int flags, AVDictionary **o
/* open the encrypted connection */
ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
rt->encrypted = 1;
+ } else if (!strcmp(proto, "rtmpsrt") || rt->rtmp_over_srt) {
+ if (rt->listen)
+ av_dict_set(opts, "mode", "listener", 1);
+ else
+ av_dict_set(opts, "mode", "caller", 1);
+ ff_url_join(buf, sizeof(buf), "ffrtmpsrt", NULL, hostname, port, "%s", path);
} else {
/* open the tcp connection */
if (port < 0)
@@ -3116,6 +3123,7 @@ static const AVOption rtmp_options[] = {
{"rtmp_listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{"listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{"timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies -rtmp_listen 1", OFFSET(listen_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
+ {"rtmp_srt", "Force RTMP over SRT", OFFSET(rtmp_over_srt), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DEC|ENC},
{ NULL },
};
@@ -3153,3 +3161,4 @@ RTMP_PROTOCOL(rtmps, RTMPS)
RTMP_PROTOCOL(rtmpt, RTMPT)
RTMP_PROTOCOL(rtmpte, RTMPTE)
RTMP_PROTOCOL(rtmpts, RTMPTS)
+RTMP_PROTOCOL(rtmpsrt, RTMPSRT)
diff --git a/libavformat/rtmpsrt.c b/libavformat/rtmpsrt.c
new file mode 100644
index 0000000000..0325973db9
--- /dev/null
+++ b/libavformat/rtmpsrt.c
@@ -0,0 +1,167 @@
+/*
+ * 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
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/intfloat.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+#include "internal.h"
+#include "url.h"
+
+typedef struct RTMP_SrtContext {
+ const AVClass *class;
+ URLContext *stream;
+ char buf[1500];
+ int buf_len;
+ int buf_index;
+ char *streamid;
+} RTMP_SrtContext;
+
+static int rtmp_srt_open(URLContext *h, const char *uri, int flags, AVDictionary **opts)
+{
+ RTMP_SrtContext *s = h->priv_data;
+ char buf[512];
+ char host[256];
+ int port;
+ char path[1024];
+ char *streamid;
+ char *p;
+
+ av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, path, sizeof(path), uri);
+
+ if (s->streamid) {
+ streamid = av_strdup(s->streamid);
+ } else {
+ // rtmp path: /${app}/{stream}?txSecret=${txSecret}&txTime=${txTime}
+ // streamid=#!::h=${rtmp-push-domain},r=${app}/${stream},txSecret=${txSecret},txTime=${txTime}
+ for (p = path; *p; p++) {
+ if (*p == '&' || *p == '?')
+ *p = ',';
+ }
+ if (path[0] == '/')
+ p = path + 1;
+ else
+ p = path;
+ streamid = av_asprintf("#!::h=%s,r=%s", host, p);
+ }
+ av_log(h, AV_LOG_DEBUG, "streamid %s\n", streamid ? streamid : "");
+ av_dict_set(opts, "streamid", streamid, AV_DICT_DONT_STRDUP_VAL);
+
+ av_dict_set(opts, "tlpktdrop", "0", 1);
+ av_dict_set(opts, "payload_size", "max_size", 1);
+
+ ff_url_join(buf, sizeof(buf), "srt", NULL, host, port, NULL);
+ return ffurl_open_whitelist(
+ &s->stream, buf, AVIO_FLAG_READ_WRITE, &h->interrupt_callback, opts,
+ h->protocol_whitelist, h->protocol_blacklist, h);
+}
+
+static int read_from_buf(RTMP_SrtContext *s, unsigned char *buf, int size)
+{
+ int min = FFMIN(s->buf_len, size);
+ memcpy(buf, s->buf + s->buf_index, min);
+ if (min == s->buf_len) {
+ s->buf_len = 0;
+ s->buf_index = 0;
+ } else {
+ s->buf_len -= min;
+ s->buf_index += min;
+ }
+ return min;
+}
+
+static int rtmp_srt_read(URLContext *h, unsigned char *buf, int size)
+{
+ int ret;
+ RTMP_SrtContext *s = h->priv_data;
+ if (s->buf_len > 0) {
+ return read_from_buf(s, buf, size);
+ }
+
+ if (h->flags & AVIO_FLAG_NONBLOCK)
+ s->stream->flags |= AVIO_FLAG_NONBLOCK;
+ else
+ s->stream->flags &= ~AVIO_FLAG_NONBLOCK;
+ ret = ffurl_read(s->stream, s->buf, s->stream->max_packet_size);
+ if (ret < 0) {
+ return ret;
+ }
+ s->buf_len = ret;
+ s->buf_index = 0;
+ return read_from_buf(s, buf, size);
+}
+
+static int rtmp_srt_write(URLContext *h, const unsigned char *buf, int size)
+{
+ int ret;
+ int n;
+ int len = 0;
+ RTMP_SrtContext *s = h->priv_data;
+
+ if (h->flags & AVIO_FLAG_NONBLOCK)
+ s->stream->flags |= AVIO_FLAG_NONBLOCK;
+ else
+ s->stream->flags &= ~AVIO_FLAG_NONBLOCK;
+ while (size > 0) {
+ n = size > s->stream->max_packet_size ? s->stream->max_packet_size : size;
+ ret = ffurl_write(s->stream, buf + len, n);
+ if (ret < 0) {
+ return ret;
+ }
+ len += ret;
+ size -= ret;
+ }
+
+ return len;
+}
+
+static int rtmp_srt_close(URLContext *h)
+{
+ RTMP_SrtContext *s = h->priv_data;
+ return ffurl_closep(&s->stream);
+}
+
+#define OFFSET(x) offsetof(RTMP_SrtContext, x)
+#define DEC AV_OPT_FLAG_DECODING_PARAM
+#define ENC AV_OPT_FLAG_ENCODING_PARAM
+
+static const AVOption ffrtmpsrt_options[] = {
+ // There is a streamid option in ffmpeg_opt. When libsrt is used by rtmp,
+ // the streamid option was passed to ffmpeg_opt and leads to error.
+ { "rtmpsrt_streamid", "A string of up to 512 characters that an Initiator can pass to a Responder", OFFSET(streamid), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = DEC|ENC },
+ { NULL },
+};
+
+static const AVClass ffrtmpsrt_class = {
+ .class_name = "ffrtmpsrt",
+ .item_name = av_default_item_name,
+ .option = ffrtmpsrt_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+const URLProtocol ff_ffrtmpsrt_protocol = {
+ .name = "ffrtmpsrt",
+ .url_open2 = rtmp_srt_open,
+ .url_read = rtmp_srt_read,
+ .url_write = rtmp_srt_write,
+ .url_close = rtmp_srt_close,
+ .priv_data_size = sizeof(RTMP_SrtContext),
+ .flags = URL_PROTOCOL_FLAG_NETWORK,
+ .priv_data_class= &ffrtmpsrt_class,
+ .default_whitelist = "srt",
+};
--
2.25.1
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