AI: Support G711 with PCMU/PCMA.

Author: ossrs-ai

trunk/research/players/js/srs.page.js

@@ -24,7 +24,7 @@ function update_nav() {
     $("#nav_vlc").attr("href", "vlc.html" + window.location.search);
 }
 
-// Special extra params, such as auth_key.
+// Special extra params, such as auth_key, codec, vcodec, acodec.
 function user_extra_params(query, params, rtc) {
     var queries = params || [];
 
@@ -124,6 +124,9 @@ function build_default_whip_whep_url(query, apiPath) {
     console.log('?api=x to overwrite WebRTC API(1985).');
     console.log('?schema=http|https to overwrite WebRTC API protocol.');
     console.log(`?path=xxx to overwrite default ${apiPath}`);
+    console.log('?codec=xxx to specify video codec (alias for vcodec, e.g., h264, vp9, av1)');
+    console.log('?vcodec=xxx to specify video codec (e.g., h264, vp9, av1)');
+    console.log('?acodec=xxx to specify audio codec (e.g., opus, pcmu, pcma)');
 
     var server = (!query.server)? window.location.hostname:query.server;
     var vhost = (!query.vhost)? window.location.hostname:query.vhost;

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

@@ -41,11 +41,15 @@ function SrsRtcWhipWhepAsync() {
     //      camera: boolean, whether capture video from camera, default to true.
     //      screen: boolean, whether capture video from screen, default to false.
     //      audio: boolean, whether play audio, default to true.
+    //      vcodec: string, video codec to use (e.g., 'h264', 'vp9', 'av1'), default to undefined.
+    //      acodec: string, audio codec to use (e.g., 'opus', 'pcmu', 'pcma'), default to undefined.
     self.publish = async function (url, options) {
         if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
         const hasAudio = options?.audio ?? true;
         const useCamera = options?.camera ?? true;
         const useScreen = options?.screen ?? false;
+        const vcodec = options?.vcodec;
+        const acodec = options?.acodec;
 
         if (!hasAudio && !useCamera && !useScreen) throw new Error(`The camera, screen and audio can't be false at the same time`);
 
@@ -91,6 +95,13 @@ function SrsRtcWhipWhepAsync() {
 
         var offer = await self.pc.createOffer();
         await self.pc.setLocalDescription(offer);
+
+        // Filter codecs if specified
+        if (vcodec || acodec) {
+            offer.sdp = self.__internal.filterCodec(offer.sdp, vcodec, acodec);
+            console.log(`Filtered codecs (vcodec=${vcodec}, acodec=${acodec}): ${offer.sdp}`);
+        }
+
         const answer = await new Promise(function (resolve, reject) {
             console.log(`Generated offer: ${offer.sdp}`);
 
@@ -119,15 +130,26 @@ function SrsRtcWhipWhepAsync() {
     // @options The options to control playing, supports:
     //      videoOnly: boolean, whether only play video, default to false.
     //      audioOnly: boolean, whether only play audio, default to false.
+    //      vcodec: string, video codec to use (e.g., 'h264', 'vp9', 'av1'), default to undefined.
+    //      acodec: string, audio codec to use (e.g., 'opus', 'pcmu', 'pcma'), default to undefined.
     self.play = async function(url, options) {
         if (url.indexOf('/whip-play/') === -1 && url.indexOf('/whep/') === -1) throw new Error(`invalid WHEP url ${url}`);
         if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
+        const vcodec = options?.vcodec;
+        const acodec = options?.acodec;
 
         if (!options?.videoOnly) self.pc.addTransceiver("audio", {direction: "recvonly"});
         if (!options?.audioOnly) self.pc.addTransceiver("video", {direction: "recvonly"});
 
         var offer = await self.pc.createOffer();
         await self.pc.setLocalDescription(offer);
+
+        // Filter codecs if specified
+        if (vcodec || acodec) {
+            offer.sdp = self.__internal.filterCodec(offer.sdp, vcodec, acodec);
+            console.log(`Filtered codecs (vcodec=${vcodec}, acodec=${acodec}): ${offer.sdp}`);
+        }
+
         const answer = await new Promise(function(resolve, reject) {
             console.log(`Generated offer: ${offer.sdp}`);
 
@@ -199,6 +221,43 @@ function SrsRtcWhipWhepAsync() {
                 simulator: a.protocol + '//' + a.host + '/rtc/v1/nack/',
             };
         },
+        filterCodec: (sdp, vcodec, acodec) => {
+            // Filter video codec if specified
+            if (vcodec) {
+                const vcodecUpper = vcodec.toUpperCase();
+                sdp = sdp.split('\n').filter(line => {
+                    // Keep all non-video lines
+                    if (!line.startsWith('a=rtpmap:') && !line.startsWith('a=rtcp-fb:') &&
+                        !line.startsWith('a=fmtp:')) {
+                        return true;
+                    }
+                    // For video codec lines, only keep the specified codec
+                    if (line.includes('video/')) {
+                        return line.toUpperCase().includes(vcodecUpper);
+                    }
+                    return true;
+                }).join('\n');
+            }
+
+            // Filter audio codec if specified
+            if (acodec) {
+                const acodecUpper = acodec.toUpperCase();
+                sdp = sdp.split('\n').filter(line => {
+                    // Keep all non-audio lines
+                    if (!line.startsWith('a=rtpmap:') && !line.startsWith('a=rtcp-fb:') &&
+                        !line.startsWith('a=fmtp:')) {
+                        return true;
+                    }
+                    // For audio codec lines, only keep the specified codec
+                    if (line.includes('audio/')) {
+                        return line.toUpperCase().includes(acodecUpper);
+                    }
+                    return true;
+                }).join('\n');
+            }
+
+            return sdp;
+        },
     };
 
     // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack

trunk/research/players/whep.html

@@ -125,9 +125,17 @@
 
         // For example: webrtc://r.ossrs.net/live/livestream
         var url = $("#txt_url").val();
+        var query = parse_query_string();
+
+        // Support codec parameters: codec (alias for vcodec), vcodec, acodec
+        var vcodec = query.vcodec || query.codec;
+        var acodec = query.acodec;
+
         sdk.play(url, {
             videoOnly: $('#ch_videoonly').prop('checked'),
             audioOnly: $('#ch_audioonly').prop('checked'),
+            vcodec: vcodec,
+            acodec: acodec
         }).then(function(session){
             $('#sessionid').html(session.sessionid);
             $('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);

trunk/research/players/whip.html

@@ -132,10 +132,18 @@
 
         // For example: webrtc://r.ossrs.net/live/livestream
         var url = $("#txt_url").val();
+        var query = parse_query_string();
+
+        // Support codec parameters: codec (alias for vcodec), vcodec, acodec
+        var vcodec = query.vcodec || query.codec;
+        var acodec = query.acodec;
+
         sdk.publish(url, {
             camera: $('#ra_camera').prop('checked'),
             screen: $('#ra_screen').prop('checked'),
-            audio: $('#ch_audio').prop('checked')
+            audio: $('#ch_audio').prop('checked'),
+            vcodec: vcodec,
+            acodec: acodec
         }).then(function(session){
             $('#sessionid').html(session.sessionid);
             $('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);

trunk/src/app/srs_app_log.cpp

@@ -20,7 +20,7 @@
 #include <srs_kernel_utility.hpp>
 
 // the max size of a line of log.
-#define LOG_MAX_SIZE 8192
+#define LOG_MAX_SIZE 65536 // 64 KB
 
 // the tail append to each log.
 #define LOG_TAIL '\n'

trunk/src/app/srs_app_rtc_api.cpp

@@ -158,19 +158,25 @@ srs_error_t SrsGoApiRtcPlay::do_serve_http(ISrsHttpResponseWriter *w, ISrsHttpMe
     if (eip.empty()) {
         eip = r->query_get("candidate");
     }
-    string codec = r->query_get("codec");
+    // Support vcodec/codec (alias for vcodec) and acodec parameters
+    string vcodec = r->query_get("vcodec");
+    if (vcodec.empty()) {
+        vcodec = r->query_get("codec");
+    }
+    string acodec = r->query_get("acodec");
     // 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, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s",
+        "RTC play %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, vcodec=%s, acodec=%s, srtp=%s, dtls=%s",
         streamurl.c_str(), api.c_str(), tid.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(), srtp.c_str(), dtls.c_str());
+        eip.c_str(), vcodec.c_str(), acodec.c_str(), srtp.c_str(), dtls.c_str());
 
     ruc.eip_ = eip;
-    ruc.codec_ = codec;
+    ruc.vcodec_ = vcodec;
+    ruc.acodec_ = acodec;
     ruc.publish_ = false;
     ruc.dtls_ = (dtls != "false");
 
@@ -479,14 +485,20 @@ srs_error_t SrsGoApiRtcPublish::do_serve_http(ISrsHttpResponseWriter *w, ISrsHtt
     if (eip.empty()) {
         eip = r->query_get("candidate");
     }
-    string codec = r->query_get("codec");
+    // Support vcodec/codec (alias for vcodec) and acodec parameters
+    string vcodec = r->query_get("vcodec");
+    if (vcodec.empty()) {
+        vcodec = r->query_get("codec");
+    }
+    string acodec = r->query_get("acodec");
 
-    srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s",
+    srs_trace("RTC publish %s, api=%s, tid=%s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, vcodec=%s, acodec=%s",
               streamurl.c_str(), api.c_str(), tid.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());
+              remote_sdp_str.length(), eip.c_str(), vcodec.c_str(), acodec.c_str());
 
     ruc.eip_ = eip;
-    ruc.codec_ = codec;
+    ruc.vcodec_ = vcodec;
+    ruc.acodec_ = acodec;
     ruc.publish_ = true;
     ruc.dtls_ = ruc.srtp_ = true;
 
@@ -776,7 +788,12 @@ srs_error_t SrsGoApiRtcWhip::do_serve_http_with(ISrsHttpResponseWriter *w, ISrsH
     if (eip.empty()) {
         eip = r->query_get("candidate");
     }
-    string codec = r->query_get("codec");
+    // Support vcodec/codec (alias for vcodec) and acodec parameters
+    string vcodec = r->query_get("vcodec");
+    if (vcodec.empty()) {
+        vcodec = r->query_get("codec");
+    }
+    string acodec = r->query_get("acodec");
     string app = r->query_get("app");
     string stream = r->query_get("stream");
     string action = r->query_get("action");
@@ -815,13 +832,14 @@ srs_error_t SrsGoApiRtcWhip::do_serve_http_with(ISrsHttpResponseWriter *w, ISrsH
     string srtp = r->query_get("encrypt");
     string dtls = r->query_get("dtls");
 
-    srs_trace("RTC whip %s %s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, codec=%s, srtp=%s, dtls=%s, ufrag=%s, pwd=%s, param=%s",
+    srs_trace("RTC whip %s %s, clientip=%s, app=%s, stream=%s, offer=%dB, eip=%s, vcodec=%s, acodec=%s, srtp=%s, dtls=%s, ufrag=%s, pwd=%s, param=%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(), srtp.c_str(), dtls.c_str(), ruc->req_->ice_ufrag_.c_str(),
+              remote_sdp_str.length(), eip.c_str(), vcodec.c_str(), acodec.c_str(), srtp.c_str(), dtls.c_str(), ruc->req_->ice_ufrag_.c_str(),
               ruc->req_->ice_pwd_.c_str(), ruc->req_->param_.c_str());
 
     ruc->eip_ = eip;
-    ruc->codec_ = codec;
+    ruc->vcodec_ = vcodec;
+    ruc->acodec_ = acodec;
     ruc->publish_ = (action == "publish");
 
     // For client to specifies whether encrypt by SRTP.

trunk/src/app/srs_app_rtc_conn.cpp

@@ -3335,10 +3335,30 @@ srs_error_t SrsRtcPublisherNegotiator::negotiate_publish_capability(SrsRtcUserCo
             // Update the ruc, which is about user specified configuration.
             ruc->audio_before_video_ = !nn_any_video_parsed;
 
-            // TODO: check opus format specific param
-            std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
+            // Try to find audio codec based on user preference or default order
+            std::vector<SrsMediaPayloadType> payloads;
+
+            // If user specified audio codec, try that first
+            if (!ruc->acodec_.empty()) {
+                payloads = remote_media_desc.find_media_with_encoding_name(ruc->acodec_);
+                if (payloads.empty()) {
+                    return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found %s audio payload type", ruc->acodec_.c_str());
+                }
+            } else {
+                // Default order: Opus, PCMU (G.711 μ-law), PCMA (G.711 A-law)
+                // Prioritize PCMU over PCMA as per Chrome SDP order
+                payloads = remote_media_desc.find_media_with_encoding_name("opus");
+                if (payloads.empty()) {
+                    // Then try PCMU (G.711 μ-law)
+                    payloads = remote_media_desc.find_media_with_encoding_name("PCMU");
+                }
+                if (payloads.empty()) {
+                    // Finally try PCMA (G.711 A-law)
+                    payloads = remote_media_desc.find_media_with_encoding_name("PCMA");
+                }
+            }
             if (payloads.empty()) {
-                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found opus payload type");
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found audio payload type (opus/PCMU/PCMA)");
             }
 
             for (int j = 0; j < (int)payloads.size(); j++) {
@@ -3366,10 +3386,10 @@ srs_error_t SrsRtcPublisherNegotiator::negotiate_publish_capability(SrsRtcUserCo
 
                 track_desc->type_ = "audio";
                 track_desc->set_codec_payload((SrsCodecPayload *)audio_payload);
-                // Only choose one match opus codec.
+                // Only choose one match audio codec.
                 break;
             }
-        } else if (remote_media_desc.is_video() && srs_video_codec_str2id(ruc->codec_) == SrsVideoCodecIdAV1) {
+        } else if (remote_media_desc.is_video() && srs_video_codec_str2id(ruc->vcodec_) == SrsVideoCodecIdAV1) {
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("AV1");
             if (payloads.empty()) {
                 // Be compatible with the Chrome M96, still check the AV1X encoding name
@@ -3406,7 +3426,7 @@ srs_error_t SrsRtcPublisherNegotiator::negotiate_publish_capability(SrsRtcUserCo
                 track_desc->set_codec_payload((SrsCodecPayload *)video_payload);
                 break;
             }
-        } else if (remote_media_desc.is_video() && srs_video_codec_str2id(ruc->codec_) == SrsVideoCodecIdVP9) {
+        } else if (remote_media_desc.is_video() && srs_video_codec_str2id(ruc->vcodec_) == SrsVideoCodecIdVP9) {
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("VP9");
             if (payloads.empty()) {
                 return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid VP9 payload type");
@@ -3438,7 +3458,7 @@ srs_error_t SrsRtcPublisherNegotiator::negotiate_publish_capability(SrsRtcUserCo
                 track_desc->set_codec_payload((SrsCodecPayload *)video_payload);
                 break;
             }
-        } else if (remote_media_desc.is_video() && srs_video_codec_str2id(ruc->codec_) == SrsVideoCodecIdHEVC) {
+        } else if (remote_media_desc.is_video() && srs_video_codec_str2id(ruc->vcodec_) == SrsVideoCodecIdHEVC) {
             std::vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("H265");
             if (payloads.empty()) {
                 return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no found valid H.265 payload type");
@@ -3797,16 +3817,33 @@ srs_error_t SrsRtcPlayerNegotiator::negotiate_play_capability(SrsRtcUserConfig *
             // Update the ruc, which is about user specified configuration.
             ruc->audio_before_video_ = !nn_any_video_parsed;
 
-            // TODO: check opus format specific param
-            vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name("opus");
+            // Try to find audio tracks in source with different codec names
+            // Try Opus first (most common), then PCMU, then PCMA
+            std::vector<SrsRtcTrackDescription *> source_audio_tracks = source->get_track_desc("audio", "opus");
+            std::string source_audio_codec = "opus";
+
+            if (source_audio_tracks.empty()) {
+                source_audio_tracks = source->get_track_desc("audio", "PCMU");
+                source_audio_codec = "PCMU";
+            }
+            if (source_audio_tracks.empty()) {
+                source_audio_tracks = source->get_track_desc("audio", "PCMA");
+                source_audio_codec = "PCMA";
+            }
+            if (source_audio_tracks.empty()) {
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no audio track in source (tried opus/PCMU/PCMA)");
+            }
+
+            // Try to find matching codec in remote SDP
+            vector<SrsMediaPayloadType> payloads = remote_media_desc.find_media_with_encoding_name(source_audio_codec);
             if (payloads.empty()) {
-                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found opus payload type");
+                return srs_error_new(ERROR_RTC_SDP_EXCHANGE, "no valid found %s payload type", source_audio_codec.c_str());
             }
 
             remote_payload = payloads.at(0);
-            track_descs = source->get_track_desc("audio", "opus");
+            track_descs = source_audio_tracks;
         } else if (remote_media_desc.is_video()) {
-            SrsVideoCodecId prefer_codec = srs_video_codec_str2id(ruc->codec_);
+            SrsVideoCodecId prefer_codec = srs_video_codec_str2id(ruc->vcodec_);
             if (prefer_codec == SrsVideoCodecIdReserved) {
                 // Get the source codec if not specified.
                 std::vector<SrsRtcTrackDescription *> source_track_descs = source->get_track_desc("video", "");

trunk/src/app/srs_app_rtc_server.hpp

@@ -70,7 +70,8 @@ class SrsRtcUserConfig
     std::string remote_sdp_str_;
     SrsSdp remote_sdp_;
     std::string eip_;
-    std::string codec_;
+    std::string vcodec_; // Video codec
+    std::string acodec_; // Audio codec
     std::string api_;
 
     // Session data.

trunk/src/app/srs_app_rtc_source.cpp

@@ -3838,7 +3838,7 @@ srs_error_t SrsRtcFormat::on_rtp_packet(SrsRtcRecvTrack *track, bool is_audio)
         SrsAudioCodecId codec_id = (SrsAudioCodecId)media->codec(false);
 
         // Parse channels and sample rate from track description
-        if (codec_id == SrsAudioCodecIdOpus) {
+        if (codec_id == SrsAudioCodecIdOpus || codec_id == SrsAudioCodecIdPCMA || codec_id == SrsAudioCodecIdPCMU) {
             SrsAudioPayload *audio_media = dynamic_cast<SrsAudioPayload *>(media);
             if (!audio_media) {
                 return err;

trunk/src/app/srs_app_statistic.cpp

@@ -178,7 +178,13 @@ srs_error_t SrsStatisticStream::dumps(SrsJsonObject *obj)
         audio->set("codec", SrsJsonAny::str(srs_audio_codec_id2str(acodec_).c_str()));
         audio->set("sample_rate", SrsJsonAny::integer(srs_audio_sample_rate2number(asample_rate_)));
         audio->set("channel", SrsJsonAny::integer(asound_type_ + 1));
-        audio->set("profile", SrsJsonAny::str(srs_aac_object2str(aac_object_).c_str()));
+
+        // G.711 codecs don't have profiles, similar to VP9/AV1
+        if (acodec_ == SrsAudioCodecIdPCMA || acodec_ == SrsAudioCodecIdPCMU) {
+            audio->set("profile", SrsJsonAny::null());
+        } else {
+            audio->set("profile", SrsJsonAny::str(srs_aac_object2str(aac_object_).c_str()));
+        }
     }
 
     return err;