For #307, support publish RTC with passing opus. 4.0.24

README.md

@@ -159,6 +159,7 @@ For previous versions, please read:
 
 ## V4 changes
 
+* v4.0, 2020-04-30, For [#307][bug #307], support publish RTC with passing opus. 4.0.24
 * v4.0, 2020-04-14, For [#307][bug #307], support sendmmsg, GSO and reuseport. 4.0.23
 * v4.0, 2020-04-05, For [#307][bug #307], SRTP ASM only works with openssl-1.0, auto detect it. 4.0.22
 * v4.0, 2020-04-04, Merge RTC and GB28181, with bugs fixed. 4.0.21

trunk/src/app/srs_app_rtc_conn.cpp

@@ -769,6 +769,7 @@ srs_error_t SrsRtcSenderThread::cycle()
 
     SrsConsumer* consumer = NULL;
     SrsAutoFree(SrsConsumer, consumer);
+    // TODO: FIXME: Dumps the SPS/PPS from gop cache, without other frames.
     if ((err = source->create_consumer(NULL, consumer)) != srs_success) {
         return srs_error_wrap(err, "rtc create consumer, source url=%s", req->get_stream_url().c_str());
     }
@@ -1967,8 +1968,26 @@ srs_error_t SrsRtcPublisher::collect_audio_frame()
 
         for (size_t j = 0; j < frames.size(); ++j) {
             SrsRtpSharedPacket* pkt = frames[j];
-            
+
-            // TODO: FIXME: Write audio frame to source.
+            if (pkt->rtp_payload_size() > 0) {
+                SrsMessageHeader header;
+                header.message_type = RTMP_MSG_AudioMessage;
+                // TODO: FIXME: Maybe the tbn is not 90k.
+                header.timestamp = pkt->rtp_header.get_timestamp() / 90;
+
+                SrsSharedPtrMessage msg;
+                // TODO: FIXME: Check error.
+                msg.create(&header, NULL, 0);
+
+                SrsSample sample;
+                sample.size = pkt->rtp_payload_size();
+                sample.bytes = new char[sample.size];
+                memcpy((void*)sample.bytes, pkt->rtp_payload(), sample.size);
+                msg.set_extra_payloads(&sample, 1);
+
+                // TODO: FIXME: Check error.
+                source->on_rtc_audio(&msg);
+            }
 
             srs_freep(pkt);
         }
@@ -2103,11 +2122,12 @@ srs_error_t SrsRtcPublisher::collect_video_frame()
             stream->write_string(pps);
 
             SrsMessageHeader header;
-            header.message_type = 9;
+            header.message_type = RTMP_MSG_VideoMessage;
             // TODO: FIXME: Maybe the tbn is not 90k.
             header.timestamp = timestamp / 90;
             SrsCommonMessage* shared_video = new SrsCommonMessage();
             SrsAutoFree(SrsCommonMessage, shared_video);
+            // TODO: FIXME: Check error.
             shared_video->create(&header, reinterpret_cast<char*>(video_header), stream->pos());
             srs_error_t e = source->on_video(shared_video);
             if (e != srs_success) {
@@ -2130,11 +2150,12 @@ srs_error_t SrsRtcPublisher::collect_video_frame()
             frame_buffer[4] = 0x00;
 
             SrsMessageHeader header;
-            header.message_type = 9;
+            header.message_type = RTMP_MSG_VideoMessage;
             // TODO: FIXME: Maybe the tbn is not 90k.
             header.timestamp = timestamp / 90;
             SrsCommonMessage* shared_video = new SrsCommonMessage();
             SrsAutoFree(SrsCommonMessage, shared_video);
+            // TODO: FIXME: Check error.
             shared_video->create(&header, reinterpret_cast<char*>(frame_buffer), frame_buffer_index);
             srs_error_t e = source->on_video(shared_video);
             if (e != srs_success) {

trunk/src/app/srs_app_source.cpp

@@ -1030,7 +1030,8 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
     srs_error_t err = srs_success;
     
     SrsSharedPtrMessage* msg = shared_audio;
-    
+
+    // TODO: FIXME: Support parsing OPUS for RTC.
     if ((err = format->on_audio(msg)) != srs_success) {
         return srs_error_wrap(err, "format consume audio");
     }
@@ -1058,6 +1059,7 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
     }
 
 #ifdef SRS_RTC
+    // TODO: FIXME: Support parsing OPUS for RTC.
     if ((err = rtc->on_audio(msg, format)) != srs_success) {
         srs_warn("rtc: ignore audio error %s", srs_error_desc(err).c_str());
         srs_error_reset(err);
@@ -2761,4 +2763,11 @@ void SrsSource::request_keyframe()
         rtc_publisher->request_keyframe();
     }
 }
+
+srs_error_t SrsSource::on_rtc_audio(SrsSharedPtrMessage* audio)
+{
+    // TODO: FIXME: Merge with on_audio.
+    // TODO: FIXME: Print key information.
+    return on_audio_imp(audio);
+}
 #endif

trunk/src/app/srs_app_source.hpp

@@ -613,10 +613,12 @@ public:
     virtual bool can_publish(bool is_edge);
     virtual srs_error_t on_meta_data(SrsCommonMessage* msg, SrsOnMetaDataPacket* metadata);
 public:
+    // TODO: FIXME: Use SrsSharedPtrMessage instead.
     virtual srs_error_t on_audio(SrsCommonMessage* audio);
 private:
     virtual srs_error_t on_audio_imp(SrsSharedPtrMessage* audio);
 public:
+    // TODO: FIXME: Use SrsSharedPtrMessage instead.
     virtual srs_error_t on_video(SrsCommonMessage* video);
 private:
     virtual srs_error_t on_video_imp(SrsSharedPtrMessage* video);
@@ -658,6 +660,7 @@ public:
     void request_keyframe();
     void set_rtc_publisher(SrsRtcPublisher* v) { rtc_publisher = v; }
     // When got RTC audio message, which is encoded in opus.
+    // TODO: FIXME: Merge with on_audio.
     srs_error_t on_rtc_audio(SrsSharedPtrMessage* audio);
 #endif
 };

trunk/src/core/srs_core_version4.hpp

@@ -24,6 +24,6 @@
 #ifndef SRS_CORE_VERSION4_HPP
 #define SRS_CORE_VERSION4_HPP
 
-#define SRS_VERSION4_REVISION 23
+#define SRS_VERSION4_REVISION 24
 
 #endif

trunk/src/kernel/srs_kernel_codec.cpp

@@ -603,7 +603,7 @@ srs_error_t SrsFormat::on_audio(int64_t timestamp, char* data, int size)
     srs_error_t err = srs_success;
     
     if (!data || size <= 0) {
-        srs_trace("no audio present, ignore it.");
+        srs_info("no audio present, ignore it.");
         return err;
     }
     

trunk/src/kernel/srs_kernel_flv.hpp

@@ -158,6 +158,7 @@ public:
     // 1byte.
     // One byte field to represent the message type. A range of type IDs
     // (1-7) are reserved for protocol control messages.
+    // For example, RTMP_MSG_AudioMessage or RTMP_MSG_VideoMessage.
     int8_t message_type;
     // 4bytes.
     // Four-byte field that identifies the stream of the message. These
@@ -245,6 +246,7 @@ public:
     // 1byte.
     // One byte field to represent the message type. A range of type IDs
     // (1-7) are reserved for protocol control messages.
+    // For example, RTMP_MSG_AudioMessage or RTMP_MSG_VideoMessage.
     int8_t message_type;
     // Get the perfered cid(chunk stream id) which sendout over.
     // set at decoding, and canbe used for directly send message,