RTC: Support av1 for Chrome M90 enabled it. 4.0.91

README.md

@@ -157,6 +157,7 @@ Other important wiki:
 
 ## V4 changes
 
+* v4.0, 2021-04-29, RTC: Support av1 for Chrome M90. 4.0.91
 * v4.0, 2021-04-24, Change push-RTSP as deprecated feature.
 * v4.0, 2021-04-24, Player: Change the default from RTMP to HTTP-FLV.
 * v4.0, 2021-04-24, Disable CherryPy by --cherrypy=off. 4.0.90

trunk/research/players/js/srs.sdk.js

@@ -470,3 +470,32 @@ function SrsRtcPlayerAsync() {
     return self;
 }
 
+// Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
+// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
+function SrsRtcFormatSenders(senders, kind) {
+    var codecs = [];
+    senders.forEach(function (sender) {
+        sender.getParameters().codecs.forEach(function(c) {
+            if (kind && sender.track.kind !== kind) {
+                return;
+            }
+
+            if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
+                return;
+            }
+
+            var s = '';
+
+            s += c.mimeType.replace('audio/', '').replace('video/', '');
+            s += ', ' + c.clockRate + 'HZ';
+            if (sender.track.kind === "audio") {
+                s += ', channels: ' + c.channels;
+            }
+            s += ', pt: ' + c.payloadType;
+
+            codecs.push(s);
+        });
+    });
+    return codecs.join(", ");
+}
+

trunk/research/players/rtc_publisher.html

@@ -55,6 +55,10 @@
     <label></label>
     SessionID: <span id='sessionid'></span>
 
+    <label></label>
+    Audio: <span id='acodecs'></span><br/>
+    Video: <span id='vcodecs'></span>
+
     <label></label>
     Simulator: <a href='#' id='simulator-drop'>Drop</a>
 
@@ -81,6 +85,14 @@
                 $('#rtc_media_player').prop('srcObject', event.stream);
             };
 
+            // https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
+            sdk.pc.onicegatheringstatechange = function (event) {
+                if (sdk.pc.iceGatheringState === "complete") {
+                    $('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
+                    $('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
+                }
+            };
+
             // For example:
             //      webrtc://r.ossrs.net/live/livestream
             var url = $("#txt_url").val();

trunk/src/app/srs_app_rtc_api.cpp

@@ -134,18 +134,20 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMe
 
     // For client to specifies the EIP of server.
     string eip = r->query_get("eip");
+    string codec = r->query_get("codec");
     // For client to specifies whether encrypt by SRTP.
     string srtp = r->query_get("encrypt");
     string dtls = r->query_get("dtls");
 
-    srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, srtp=%s, dtls=%s",
+    srs_trace("RTC play %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s",
         streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(),
-        srtp.c_str(), dtls.c_str()
+        codec.c_str(), srtp.c_str(), dtls.c_str()
     );
 
     // The RTC user config object.
     SrsRtcUserConfig ruc;
     ruc.eip_ = eip;
+    ruc.codec_ = codec;
     ruc.publish_ = false;
     ruc.dtls_ = (dtls != "false");
 
@@ -500,14 +502,17 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter* w, ISrsHtt
 
     // For client to specifies the EIP of server.
     string eip = r->query_get("eip");
+    string codec = r->query_get("codec");
 
-    srs_trace("RTC publish %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s",
-        streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str()
+    srs_trace("RTC publish %s, api=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
+        streamurl.c_str(), api.c_str(), clientip.c_str(), app.c_str(), stream_name.c_str(), remote_sdp_str.length(), eip.c_str(),
+        codec.c_str()
     );
 
     // The RTC user config object.
     SrsRtcUserConfig ruc;
     ruc.eip_ = eip;
+    ruc.codec_ = codec;
     ruc.publish_ = true;
     ruc.dtls_ = ruc.srtp_ = true;
 

trunk/src/app/srs_app_rtc_conn.cpp

@@ -2771,6 +2771,38 @@ srs_error_t SrsRtcConnection::negotiate_publish_capability(SrsRtcUserConfig* ruc
                 // Only choose one match opus codec.
                 break;
             }
+        } else if (remote_media_desc.is_video() && ruc->codec_ == "av1") {
+            std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("AV1X");
+            if (payloads.empty()) {
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid AV1 payload type");
+            }
+
+            for (int j = 0; j < (int)payloads.size(); j++) {
+                const SrsMediaPayloadType& payload = payloads.at(j);
+
+                // Generate video payload for av1.
+                SrsVideoPayload* video_payload = new SrsVideoPayload(payload.payload_type_, payload.encoding_name_, payload.clock_rate_);
+
+                // 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);
+
+                    if (nack_enabled) {
+                        if (rtcp_fb == "nack" || rtcp_fb == "nack pli") {
+                            video_payload->rtcp_fbs_.push_back(rtcp_fb);
+                        }
+                    }
+                    if (twcc_enabled && remote_twcc_id) {
+                        if (rtcp_fb == "transport-cc") {
+                            video_payload->rtcp_fbs_.push_back(rtcp_fb);
+                        }
+                    }
+                }
+
+                track_desc->type_ = "video";
+                track_desc->set_codec_payload((SrsCodecPayload*)video_payload);
+                break;
+            }
         } else if (remote_media_desc.is_video()) {
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");
             if (payloads.empty()) {
@@ -3050,6 +3082,14 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRtcUserConfig* ruc, s
 
             remote_payload = payloads.at(0);
             track_descs = source->get_track_desc("audio", "opus");
+        } else if (remote_media_desc.is_video() && ruc->codec_ == "av1") {
+            std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("AV1X");
+            if (payloads.empty()) {
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid AV1 payload type");
+            }
+
+            remote_payload = payloads.at(0);
+            track_descs = source->get_track_desc("video", "AV1X");
         } else if (remote_media_desc.is_video()) {
             // TODO: check opus format specific param
             vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H264");

trunk/src/app/srs_app_rtc_server.hpp

@@ -92,6 +92,7 @@ class SrsRtcUserConfig
     // Original variables from API.
     SrsSdp remote_sdp_;
     std::string eip_;
+    std::string codec_;
 
     // Generated data.
     SrsRequest* req_;

trunk/src/core/srs_core_version4.hpp

@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    90
+#define VERSION_REVISION    91
 
 #endif