Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
@@ skipping 1067 matching lines @@
             send_config->rtp.ulpfec.ulpfec_payload_type;
         (*receive_configs)[0].rtp.ulpfec.red_payload_type =
             send_config->rtp.ulpfec.red_payload_type;
         (*receive_configs)[0].rtp.ulpfec.red_rtx_payload_type =
             send_config->rtp.ulpfec.red_rtx_payload_type;
       }
 
       if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
-        (*receive_configs)[0].rtp.rtx[payload_type_].ssrc = kSendRtxSsrcs[0];
-        (*receive_configs)[0].rtp.rtx[payload_type_].payload_type =
+        (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
+        (*receive_configs)[0].rtp.rtx_payload_types[payload_type_] =
             kSendRtxPayloadType;
       }
       // Configure encoding and decoding with VP8, since generic packetization
       // doesn't support FEC with NACK.
       RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
       send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "VP8";
       (*receive_configs)[0].decoders[0].payload_name = "VP8";
     }
 
@@ skipping 1221 matching lines @@
         send_config->encoder_settings.payload_name = "VP8";
         (*receive_configs)[0].decoders[0].payload_name = "VP8";
         (*receive_configs)[0].rtp.ulpfec.red_payload_type = kRedPayloadType;
         (*receive_configs)[0].rtp.ulpfec.ulpfec_payload_type =
             kUlpfecPayloadType;
       }
       // RTX
       if (use_rtx_) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
-        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
-            kSendRtxSsrcs[0];
-        (*receive_configs)[0].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
+        (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
+        (*receive_configs)[0].rtp.rtx_payload_types[kFakeVideoSendPayloadType] =
             kSendRtxPayloadType;
       }
       // RTT needed for RemoteNtpTimeEstimator for the receive stream.
       (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
       encoder_config->content_type =
           screenshare_ ? VideoEncoderConfig::ContentType::kScreen
                        : VideoEncoderConfig::ContentType::kRealtimeVideo;
     }
 
     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
@@ skipping 599 matching lines @@
 
       const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
       for (size_t i = 0; i < ssrcs.size(); ++i) {
         expected_send_ssrcs_.insert(ssrcs[i]);
         expected_receive_ssrcs_.push_back(
             (*receive_configs)[i].rtp.remote_ssrc);
         (*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
         (*receive_configs)[i].renderer = &receive_stream_renderer_;
         (*receive_configs)[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
 
-        (*receive_configs)[i].rtp.rtx[kFakeVideoSendPayloadType].ssrc =
-            kSendRtxSsrcs[i];
-        (*receive_configs)[i].rtp.rtx[kFakeVideoSendPayloadType].payload_type =
+        (*receive_configs)[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
+        (*receive_configs)[i].rtp.rtx_payload_types[kFakeVideoSendPayloadType] =
             kSendRtxPayloadType;
       }
 
       for (size_t i = 0; i < kNumSsrcs; ++i)
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
 
       // Use a delayed encoder to make sure we see CpuOveruseMetrics stats that
       // are non-zero.
       send_config->encoder_settings.encoder = &encoder_;
     }
@@ skipping 967 matching lines @@
 
 TEST_P(EndToEndTest, VerifyDefaultVideoReceiveConfigParameters) {
   VideoReceiveStream::Config default_receive_config(nullptr);
   EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode)
       << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
   EXPECT_FALSE(default_receive_config.rtp.remb)
       << "REMB require rtcp-fb: goog-remb to be negotiated.";
   EXPECT_FALSE(
       default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
       << "RTCP XR settings require rtcp-xr to be negotiated.";
-  EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
+  EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc)
+      << "Enabling RTX requires ssrc-group: FID negotiation";
+  EXPECT_TRUE(default_receive_config.rtp.rtx_payload_types.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyUlpfecConfig(default_receive_config.rtp.ulpfec);
 }
 
 TEST_P(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
   test::NullTransport rtcp_send_transport;
@@ skipping 173 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc