Hi,
from my understand "ffrtmpsrt" combines rtmp (with flv (de-)muxing),
which demands by design for a transmission without packet loss, with a
transmission protocol based on UDP, which can have packet loss. So, I am
not sure if we actually should add something like this to our source code.
Best regards,
Thomas.
On 18.05.2021 18:03, Zhao Zhili wrote:
> ---
> 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",
> +};
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