Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.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.
 */
 
@@ skipping 246 matching lines @@
   }
 
   void RunRtxTest(RtxMode rtx_method, int loss) {
     rtp_payload_registry_.SetRtxSsrc(kTestSsrc + 1);
     rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
     rtp_rtcp_module_->SetRtxSsrc(kTestSsrc + 1);
     transport_.DropEveryNthPacket(loss);
     uint32_t timestamp = 3000;
     uint16_t nack_list[kVideoNackListSize];
     for (int frame = 0; frame < kNumFrames; ++frame) {
-      EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(
-                       webrtc::kVideoFrameDelta, kPayloadType, timestamp,
-                       timestamp / 90, payload_data, payload_data_length));
+      EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
+          webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
+          payload_data, payload_data_length, nullptr, nullptr, nullptr));
       // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
       fake_clock.AdvanceTimeMilliseconds(5);
       int length = BuildNackList(nack_list);
       if (length > 0)
         rtp_rtcp_module_->SendNACK(nack_list, length);
       fake_clock.AdvanceTimeMilliseconds(28);  //  33ms - 5ms delay.
       rtp_rtcp_module_->Process();
       // Prepare next frame.
       timestamp += 3000;
     }
@@ skipping 23 matching lines @@
   // Disable StorePackets to be able to set a larger packet history.
   rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
   // Enable StorePackets with a packet history of 2000 packets.
   rtp_rtcp_module_->SetStorePacketsStatus(true, 2000);
   // Drop 900 packets from the second one so that we get a NACK list which is
   // big enough to require 4 RTCP packets to be fully transmitted to the sender.
   transport_.DropConsecutivePackets(2, kNumPacketsToDrop);
   // Send 30 frames which at the default size is roughly what we need to get
   // enough packets.
   for (int frame = 0; frame < kNumFrames; ++frame) {
-    EXPECT_EQ(0, rtp_rtcp_module_->SendOutgoingData(
-                     webrtc::kVideoFrameDelta, kPayloadType, timestamp,
-                     timestamp / 90, payload_data, payload_data_length));
+    EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
+        webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
+        payload_data, payload_data_length, nullptr, nullptr, nullptr));
     // Prepare next frame.
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
     rtp_rtcp_module_->Process();
   }
   EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
   EXPECT_FALSE(receiver_.sequence_numbers_.empty());
   size_t last_receive_count = receiver_.sequence_numbers_.size();
   int length = BuildNackList(nack_list);
   for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
@@ skipping 11 matching lines @@
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
             *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
 
 }  // namespace webrtc