For #3174: WebRTC: Support Unity to publish or play stream. v4.0.264

ossrs · Sep 9, 2022 · aea2bfb · aea2bfb
1 parent 8ac8ae1
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Showing 7 changed files with 31 additions and 13 deletions.
diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
@@ -8,6 +8,7 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2022-09-09, For [#3174](https://github.com/ossrs/srs/issues/3174): WebRTC: Support Unity to publish or play stream. v4.0.264
 * v4.0, 2022-09-09, Fix [#3093](https://github.com/ossrs/srs/issues/3093): WebRTC: Ignore unknown fmtp for h.264. v4.0.263
 * v4.0, 2022-09-06, Fix [#3170](https://github.com/ossrs/srs/issues/3170): WebRTC: Support WHIP(WebRTC-HTTP ingestion protocol). v4.0.262
 * v4.0, 2022-09-03, Fix HTTP url parsing bug. v4.0.261

diff --git a/trunk/src/app/srs_app_rtc_api.cpp b/trunk/src/app/srs_app_rtc_api.cpp
@@ -577,11 +577,13 @@ srs_error_t SrsGoApiRtcPublish::http_hooks_on_publish(SrsRequest* req)
 SrsGoApiRtcWhip::SrsGoApiRtcWhip(SrsRtcServer* server)
 {
     publish_ = new SrsGoApiRtcPublish(server);
+    play_ = new SrsGoApiRtcPlay(server);
 }
 
 SrsGoApiRtcWhip::~SrsGoApiRtcWhip()
 {
     srs_freep(publish_);
+    srs_freep(play_);
 }
 
 srs_error_t SrsGoApiRtcWhip::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
@@ -614,6 +616,13 @@ srs_error_t SrsGoApiRtcWhip::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
     string codec = r->query_get("codec");
     string app = r->query_get("app");
     string stream = r->query_get("stream");
+    string action = r->query_get("action");
+    if (action.empty()) {
+        action = "publish";
+    }
+    if (srs_string_ends_with(r->path(), "/whip-play/")) {
+        action = "play";
+    }
 
     // The RTC user config object.
     SrsRtcUserConfig ruc;
@@ -629,14 +638,14 @@ srs_error_t SrsGoApiRtcWhip::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
         ruc.req_->vhost = parsed_vhost->arg0();
     }
 
-    srs_trace("RTC whip %s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
-        ruc.req_->get_stream_url().c_str(), clientip.c_str(), ruc.req_->app.c_str(), ruc.req_->stream.c_str(),
+    srs_trace("RTC whip %s %s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
+        action.c_str(), ruc.req_->get_stream_url().c_str(), clientip.c_str(), ruc.req_->app.c_str(), ruc.req_->stream.c_str(),
         remote_sdp_str.length(), eip.c_str(), codec.c_str()
     );
 
     ruc.eip_ = eip;
     ruc.codec_ = codec;
-    ruc.publish_ = true;
+    ruc.publish_ = (action == "publish");
     ruc.dtls_ = ruc.srtp_ = true;
 
     // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
@@ -645,7 +654,8 @@ srs_error_t SrsGoApiRtcWhip::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessa
         return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
     }
 
-    if ((err = publish_->serve_http(w, r, &ruc)) != srs_success) {
+    err = action == "publish" ? publish_->serve_http(w, r, &ruc) : play_->serve_http(w, r, &ruc);
+    if (err != srs_success) {
         return srs_error_wrap(err, "serve");
     }
 

diff --git a/trunk/src/app/srs_app_rtc_api.hpp b/trunk/src/app/srs_app_rtc_api.hpp
@@ -27,7 +27,9 @@ class SrsGoApiRtcPlay : public ISrsHttpHandler
     virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r);
 private:
     virtual srs_error_t do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsJsonObject* res);
+public:
     virtual srs_error_t serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r, SrsRtcUserConfig* ruc);
+private:
     srs_error_t check_remote_sdp(const SrsSdp& remote_sdp);
 private:
     virtual srs_error_t http_hooks_on_play(SrsRequest* req);
@@ -56,8 +58,8 @@ class SrsGoApiRtcPublish : public ISrsHttpHandler
 class SrsGoApiRtcWhip : public ISrsHttpHandler
 {
 private:
-    SrsRtcServer* server_;
     SrsGoApiRtcPublish* publish_;
+    SrsGoApiRtcPlay* play_;
 public:
     SrsGoApiRtcWhip(SrsRtcServer* server);
     virtual ~SrsGoApiRtcWhip();

diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -2977,6 +2977,9 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
                     SrsVideoPayload* video_payload = new SrsVideoPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_);
                     video_payload->set_h264_param_desc(payload.format_specific_param_);
 
+                    // Set the codec parameter for H.264, to make Unity happy.
+                    video_payload->h264_param_ = h264_param;
+
                     // TODO: FIXME: Only support some transport algorithms.
                     for (int k = 0; k < (int)payload.rtcp_fb_.size(); ++k) {
                         const string& rtcp_fb = payload.rtcp_fb_.at(k);
@@ -3061,6 +3064,8 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
                     stream_desc->audio_track_desc_ = track_desc_copy;
                 } else if (remote_media_desc.is_video()) {
                     stream_desc->video_track_descs_.push_back(track_desc_copy);
+                } else {
+                    srs_freep(track_desc_copy);
                 }
             }
             track_id = ssrc_info.msid_tracker_;

diff --git a/trunk/src/app/srs_app_rtc_server.cpp b/trunk/src/app/srs_app_rtc_server.cpp
@@ -505,10 +505,16 @@ srs_error_t SrsRtcServer::listen_api()
         return srs_error_wrap(err, "handle publish");
     }
 
+    // Generally, WHIP is a publishing protocol, but it can be also used as playing.
     if ((err = http_api_mux->handle("/rtc/v1/whip/", new SrsGoApiRtcWhip(this))) != srs_success) {
         return srs_error_wrap(err, "handle whip");
     }
 
+    // We create another mount, to support play with the same query string as publish.
+    if ((err = http_api_mux->handle("/rtc/v1/whip-play/", new SrsGoApiRtcWhip(this))) != srs_success) {
+        return srs_error_wrap(err, "handle whip play");
+    }
+
 #ifdef SRS_SIMULATOR
     if ((err = http_api_mux->handle("/rtc/v1/nack/", new SrsGoApiRtcNACK(this))) != srs_success) {
         return srs_error_wrap(err, "handle nack");

diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
@@ -357,13 +357,7 @@ class SrsCodecPayload
 class SrsVideoPayload : public SrsCodecPayload
 {
 public:
-    struct H264SpecificParameter
-    {
-        std::string profile_level_id;
-        std::string packetization_mode;
-        std::string level_asymmerty_allow;
-    };
-    H264SpecificParameter h264_param_;
+    H264SpecificParam h264_param_;
 
 public:
     SrsVideoPayload();

diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    263
+#define VERSION_REVISION    264
 
 #endif