diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md index 41b538604..22cb37ee6 100644 --- a/trunk/doc/CHANGELOG.md +++ b/trunk/doc/CHANGELOG.md @@ -65,6 +65,8 @@ 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 * v4.0, 2022-09-03, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Play stucked when republish. v4.0.260 diff --git a/trunk/src/app/srs_app_rtc_api.cpp b/trunk/src/app/srs_app_rtc_api.cpp index 2b5882212..051b4ca6c 100644 --- a/trunk/src/app/srs_app_rtc_api.cpp +++ b/trunk/src/app/srs_app_rtc_api.cpp @@ -580,11 +580,13 @@ 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) @@ -617,6 +619,13 @@ 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; @@ -632,14 +641,14 @@ 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. @@ -648,7 +657,8 @@ 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 index 88d43491b..52c927468 100644 --- a/trunk/src/app/srs_app_rtc_api.hpp +++ b/trunk/src/app/srs_app_rtc_api.hpp @@ -27,7 +27,9 @@ public: 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 @@ private: 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 index b85dfabe7..8ec4664cb 100644 --- a/trunk/src/app/srs_app_rtc_conn.cpp +++ b/trunk/src/app/srs_app_rtc_conn.cpp @@ -2693,6 +2693,9 @@ 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); @@ -2742,6 +2745,7 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc } } + track_desc->type_ = "video"; track_desc->set_codec_payload((SrsCodecPayload*)video_payload); srs_warn("choose backup H.264 payload type=%d", payload.payload_type_); } @@ -2750,6 +2754,11 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc //local_media_desc.payload_types_.back().rtcp_fb_.push_back("rrtr"); } + // Error if track desc is invalid, that is failed to match SDP, for example, we require H264 but no H264 found. + if (track_desc->type_.empty() || !track_desc->media_) { + return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no match for track=%s, mid=%s, tracker=%s", remote_media_desc.type_.c_str(), remote_media_desc.mid_.c_str(), remote_media_desc.msid_tracker_.c_str()); + } + // TODO: FIXME: use one parse payload from sdp. track_desc->create_auxiliary_payload(remote_media_desc.find_media_with_encoding_name("red")); @@ -2771,6 +2780,8 @@ 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_sdp.cpp b/trunk/src/app/srs_app_rtc_sdp.cpp index c952b62a7..e74602bcd 100644 --- a/trunk/src/app/srs_app_rtc_sdp.cpp +++ b/trunk/src/app/srs_app_rtc_sdp.cpp @@ -56,32 +56,39 @@ static void skip_first_spaces(std::string& str) srs_error_t srs_parse_h264_fmtp(const std::string& fmtp, H264SpecificParam& h264_param) { srs_error_t err = srs_success; - std::vector vec = split_str(fmtp, ";"); + + std::vector vec = srs_string_split(fmtp, ";"); for (size_t i = 0; i < vec.size(); ++i) { - std::vector kv = split_str(vec[i], "="); - if (kv.size() == 2) { - if (kv[0] == "profile-level-id") { - h264_param.profile_level_id = kv[1]; - } else if (kv[0] == "packetization-mode") { - // 6.3. Non-Interleaved Mode - // This mode is in use when the value of the OPTIONAL packetization-mode - // media type parameter is equal to 1. This mode SHOULD be supported. - // It is primarily intended for low-delay applications. Only single NAL - // unit packets, STAP-As, and FU-As MAY be used in this mode. STAP-Bs, - // MTAPs, and FU-Bs MUST NOT be used. The transmission order of NAL - // units MUST comply with the NAL unit decoding order. - // @see https://tools.ietf.org/html/rfc6184#section-6.3 - h264_param.packetization_mode = kv[1]; - } else if (kv[0] == "level-asymmetry-allowed") { - h264_param.level_asymmerty_allow = kv[1]; - } else { - return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", kv[0].c_str()); - } - } else { - return srs_error_new(ERROR_RTC_SDP_DECODE, "invalid h264 param=%s", vec[i].c_str()); + std::vector kv = srs_string_split(vec[i], "="); + if (kv.size() != 2) continue; + + if (kv[0] == "profile-level-id") { + h264_param.profile_level_id = kv[1]; + } else if (kv[0] == "packetization-mode") { + // 6.3. Non-Interleaved Mode + // This mode is in use when the value of the OPTIONAL packetization-mode + // media type parameter is equal to 1. This mode SHOULD be supported. + // It is primarily intended for low-delay applications. Only single NAL + // unit packets, STAP-As, and FU-As MAY be used in this mode. STAP-Bs, + // MTAPs, and FU-Bs MUST NOT be used. The transmission order of NAL + // units MUST comply with the NAL unit decoding order. + // @see https://tools.ietf.org/html/rfc6184#section-6.3 + h264_param.packetization_mode = kv[1]; + } else if (kv[0] == "level-asymmetry-allowed") { + h264_param.level_asymmerty_allow = kv[1]; } } + if (h264_param.profile_level_id.empty()) { + return srs_error_new(ERROR_RTC_SDP_DECODE, "no h264 param: profile-level-id"); + } + if (h264_param.packetization_mode.empty()) { + return srs_error_new(ERROR_RTC_SDP_DECODE, "no h264 param: packetization-mode"); + } + if (h264_param.level_asymmerty_allow.empty()) { + return srs_error_new(ERROR_RTC_SDP_DECODE, "no h264 param: level-asymmetry-allowed"); + } + return err; } diff --git a/trunk/src/app/srs_app_rtc_server.cpp b/trunk/src/app/srs_app_rtc_server.cpp index f6eb3c0c9..824e3e7ff 100644 --- a/trunk/src/app/srs_app_rtc_server.cpp +++ b/trunk/src/app/srs_app_rtc_server.cpp @@ -469,10 +469,16 @@ 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 index 775a42c6d..4d52525cb 100644 --- a/trunk/src/app/srs_app_rtc_source.hpp +++ b/trunk/src/app/srs_app_rtc_source.hpp @@ -357,13 +357,7 @@ public: 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 index 1923ac488..9a8cd8ee9 100644 --- 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 262 +#define VERSION_REVISION 264 #endif