For #1638, #307, config to discard aac for rtc

trunk/conf/full.conf

@@ -335,6 +335,11 @@ vhost rtc.vhost.srs.com {
         #       discard     Discard bframe, maybe cause browser with little problems.
         # default: keep
         bframe      discard;
+        # The strategy for aac audio.
+        #       transcode   Transcode aac to opus.
+        #       discard     Discard aac audio packet.
+        # default: transcode
+        aac         transcode;
     }
 }
 

trunk/src/app/srs_app_config.cpp

@@ -3822,8 +3822,8 @@ srs_error_t SrsConfig::check_normal_config()
             } else if (n == "rtc") {
                 for (int j = 0; j < (int)conf->directives.size(); j++) {
                     string m = conf->at(j)->name;
-                    if (m != "enabled" && m != "bframe") {
+                    if (m != "enabled" && m != "bframe" && m != "aac") {
-                        return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.bandcheck.%s of %s", m.c_str(), vhost->arg0().c_str());
+                        return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
                     }
                 }
             }
@@ -4381,6 +4381,24 @@ bool SrsConfig::get_rtc_bframe_discard(string vhost)
     return conf->arg0() == "discard";
 }
 
+bool SrsConfig::get_rtc_aac_discard(string vhost)
+{
+    static bool DEFAULT = false;
+
+    SrsConfDirective* conf = get_rtc(vhost);
+
+    if (!conf) {
+        return DEFAULT;
+    }
+
+    conf = conf->get("aac");
+    if (!conf || conf->arg0().empty()) {
+        return DEFAULT;
+    }
+
+    return conf->arg0() == "discard";
+}
+
 SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
 {
     srs_assert(root);

trunk/src/app/srs_app_config.hpp

@@ -509,6 +509,7 @@ public:
     SrsConfDirective* get_rtc(std::string vhost);
     bool get_rtc_enabled(std::string vhost);
     bool get_rtc_bframe_discard(std::string vhost);
+    bool get_rtc_aac_discard(std::string vhost);
 
 // vhost specified section
 public:

trunk/src/app/srs_app_rtc.cpp

@@ -430,6 +430,8 @@ SrsRtc::SrsRtc()
     enabled = false;
     disposable = false;
     last_update_time = 0;
+
+    discard_aac = false;
 }
 
 SrsRtc::~SrsRtc()
@@ -461,6 +463,8 @@ srs_error_t SrsRtc::initialize(SrsOriginHub* h, SrsRequest* r)
 
     rtp_h264_muxer = new SrsRtpMuxer();
     rtp_h264_muxer->discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
+    // TODO: FIXME: Support reload and log it.
+    discard_aac = _srs_config->get_rtc_aac_discard(req->vhost);
 
     rtp_opus_muxer = new SrsRtpOpusMuxer();
     if (rtp_opus_muxer) {
@@ -528,6 +532,11 @@ srs_error_t SrsRtc::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* forma
         return err;
     }
 
+    // When drop aac audio packet, never transcode.
+    if (discard_aac && acodec == SrsAudioCodecIdAAC) {
+        return err;
+    }
+    
     // ignore sequence header
     srs_assert(format->audio);
 

trunk/src/app/srs_app_rtc.hpp

@@ -82,6 +82,7 @@ private:
     srs_error_t packet_stap_a(const std::string &sps, const std::string& pps, SrsSharedPtrMessage* shared_frame, std::vector<SrsRtpSharedPacket*>& rtp_packet_vec);
 };
 
+// TODO: FIXME: It's not a muxer, but a transcoder.
 class SrsRtpOpusMuxer
 {
 private:
@@ -105,6 +106,7 @@ private:
     SrsRequest* req;
     bool enabled;
     bool disposable;
+    bool discard_aac;
     srs_utime_t last_update_time;
     SrsRtpMuxer* rtp_h264_muxer;
     SrsRtpOpusMuxer* rtp_opus_muxer;