Revert of Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  Copyright (c) 2012 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 153 matching lines @@
 
   // Test detection at the end of a DTMF tone.
   // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
 
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
   // Start basic RTP test.
 
   // Send an empty RTP packet.
   // Should fail since we have not registered the payload type.
-  EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
-                                         nullptr, 0, nullptr, nullptr,
-                                         nullptr));
+  EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
+                                          96, 0, -1, NULL, 0));
 
   CodecInst voice_codec;
   memset(&voice_codec, 0, sizeof(voice_codec));
   voice_codec.pltype = 96;
   voice_codec.plfreq = 8000;
   memcpy(voice_codec.plname, "PCMU", 5);
 
   EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
   EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
       voice_codec.plname,
       voice_codec.pltype,
       voice_codec.plfreq,
       voice_codec.channels,
       (voice_codec.rate < 0) ? 0 : voice_codec.rate));
   EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
   voice_codec.rate = test_rate;
   EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
       voice_codec.plname,
       voice_codec.pltype,
       voice_codec.plfreq,
       voice_codec.channels,
       (voice_codec.rate < 0) ? 0 : voice_codec.rate));
 
   const uint8_t test[5] = "test";
-  EXPECT_EQ(true,
-            module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
-                                      test, 4, nullptr, nullptr, nullptr));
+  EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+                                         0, -1, test, 4));
 
   EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
   uint32_t timestamp;
   EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
   EXPECT_EQ(test_timestamp, timestamp);
 }
 
 TEST_F(RtpRtcpAudioTest, RED) {
   CodecInst voice_codec;
   memset(&voice_codec, 0, sizeof(voice_codec));
@@ skipping 52 matching lines @@
   fragmentation.fragmentationOffset[1] = 4;
   fragmentation.fragmentationTimeDiff = new uint16_t[2];
   fragmentation.fragmentationTimeDiff[0] = 0;
   fragmentation.fragmentationTimeDiff[1] = 0;
   fragmentation.fragmentationPlType = new uint8_t[2];
   fragmentation.fragmentationPlType[0] = 96;
   fragmentation.fragmentationPlType[1] = 96;
 
   const uint8_t test[5] = "test";
   // Send a RTP packet.
-  EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 160, -1,
-                                        test, 4, &fragmentation, nullptr,
-                                        nullptr));
+  EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
+                                         96, 160, -1, test, 4,
+                                         &fragmentation));
 
   EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
   EXPECT_EQ(-1, module1->SendREDPayloadType(&red));
 }
 
 TEST_F(RtpRtcpAudioTest, DTMF) {
   CodecInst voice_codec;
   memset(&voice_codec, 0, sizeof(voice_codec));
   voice_codec.pltype = 96;
   voice_codec.plfreq = 8000;
@@ skipping 39 matching lines @@
   for (int i = 0; i < 16; i++) {
     EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
   }
   timeStamp += 160;  // Prepare for next packet.
 
   const uint8_t test[9] = "test";
 
   // Send RTP packets for 16 tones a 160 ms  100ms
   // pause between = 2560ms + 1600ms = 4160ms
   for (; timeStamp <= 250 * 160; timeStamp += 160) {
-    EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
-                                          timeStamp, -1, test, 4, nullptr,
-                                          nullptr, nullptr));
+    EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+                                           timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
 
   for (; timeStamp <= 740 * 160; timeStamp += 160) {
-    EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
-                                          timeStamp, -1, test, 4, nullptr,
-                                          nullptr, nullptr));
+    EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
+                                           timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc