Add EncodedImageCallback::OnEncodedImage().

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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 188 matching lines @@
     const uint8_t kPayload[] = {47, 11, 32, 93, 89};
     const uint32_t kTimestamp = 1234;
     const uint8_t kPayloadType = 127;
     const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
     char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
     EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
                                               0, 1500));
 
     EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
                      kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
-                     kPayload, sizeof(kPayload), nullptr));
+                     kPayload, sizeof(kPayload), nullptr, nullptr, nullptr));
   }
 };
 
 // TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
 // default code path.
 class RtpSenderTestWithoutPacer : public RtpSenderTest {
  public:
   void SetUp() override { SetUpRtpSender(false); }
 };
 
@@ skipping 882 matching lines @@
 }
 
 TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   // Send keyframe
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
   uint8_t generic_header = *payload_data++;
 
   ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
 
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
 
   EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
 
   // Send delta frame
   payload[0] = 13;
   payload[1] = 42;
   payload[4] = 13;
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
-                                             1234, 4321, payload,
-                                             sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                   kVideoFrameDelta, payload_type, 1234, 4321, payload,
+                   sizeof(payload), nullptr, nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
                                           transport_.last_sent_packet_len_);
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_);
   generic_header = *payload_data++;
 
   EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
@@ skipping 30 matching lines @@
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
       .Times(::testing::AtLeast(2));
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr, nullptr));
 
   EXPECT_EQ(1U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(0, callback.frame_counts_.delta_frames);
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
-                                             1234, 4321, payload,
-                                             sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                   kVideoFrameDelta, payload_type, 1234, 4321, payload,
+                   sizeof(payload), nullptr, nullptr, nullptr));
 
   EXPECT_EQ(2U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(1, callback.frame_counts_.delta_frames);
 
   rtp_sender_.reset();
 }
 
 TEST_F(RtpSenderTest, BitrateCallbacks) {
@@ skipping 41 matching lines @@
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   // Initial process call so we get a new time window.
   rtp_sender_->ProcessBitrate();
 
   // Send a few frames.
   for (uint32_t i = 0; i < kNumPackets; ++i) {
-    ASSERT_EQ(0,
-              rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                            4321, payload, sizeof(payload), 0));
+    ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                     kVideoFrameKey, payload_type, 1234, 4321, payload,
+                     sizeof(payload), nullptr, nullptr, nullptr));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
   rtp_sender_->ProcessBitrate();
 
   // We get one call for every stats updated, thus two calls since both the
   // stream stats and the retransmit stats are updated once.
   EXPECT_EQ(2u, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   const uint32_t kTotalPacketSize = kPacketOverhead + sizeof(payload);
@@ skipping 58 matching lines @@
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   rtp_sender_->RegisterRtpStatisticsCallback(&callback);
 
   // Send a frame.
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr, nullptr));
   StreamDataCounters expected;
   expected.transmitted.payload_bytes = 6;
   expected.transmitted.header_bytes = 12;
   expected.transmitted.padding_bytes = 0;
   expected.transmitted.packets = 1;
   expected.retransmitted.payload_bytes = 0;
   expected.retransmitted.header_bytes = 0;
   expected.retransmitted.padding_bytes = 0;
   expected.retransmitted.packets = 0;
   expected.fec.packets = 0;
@@ skipping 19 matching lines @@
   expected.transmitted.packets = 3;
   callback.Matches(ssrc, expected);
 
   // Send FEC.
   rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType);
   FecProtectionParams fec_params;
   fec_params.fec_mask_type = kFecMaskRandom;
   fec_params.fec_rate = 1;
   fec_params.max_fec_frames = 1;
   rtp_sender_->SetFecParameters(&fec_params, &fec_params);
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
-                                             1234, 4321, payload,
-                                             sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                   kVideoFrameDelta, payload_type, 1234, 4321, payload,
+                   sizeof(payload), nullptr, nullptr, nullptr));
   expected.transmitted.payload_bytes = 40;
   expected.transmitted.header_bytes = 60;
   expected.transmitted.packets = 5;
   expected.fec.packets = 1;
   callback.Matches(ssrc, expected);
 
   rtp_sender_->RegisterRtpStatisticsCallback(nullptr);
 }
 
 TEST_F(RtpSenderAudioTest, SendAudio) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
 
   EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
 }
 
 TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
                                                        kAudioLevelExtensionId));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
@@ skipping 30 matching lines @@
   ASSERT_EQ(
       0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
   // During start, it takes the starting timestamp as last sent timestamp.
   // The duration is calculated as the difference of current and last sent
   // timestamp. So for first call it will skip since the duration is zero.
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms, 0, nullptr, 0,
-                                             nullptr));
+                                             nullptr, nullptr, nullptr));
   // DTMF Sample Length is (Frequency/1000) * Duration.
   // So in this case, it is (8000/1000) * 500 = 4000.
   // Sending it as two packets.
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 2000, 0, nullptr,
-                                             0, nullptr));
+                                             0, nullptr, nullptr, nullptr));
   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
                                 transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(rtp_header.markerBit);
 
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 4000, 0, nullptr,
-                                             0, nullptr));
+                                             0, nullptr, nullptr, nullptr));
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
                                 transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(rtp_header.markerBit);
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   const char* kPayloadName = "GENERIC";
   const uint8_t kPayloadType = 127;
   rtp_sender_->SetSSRC(1234);
   rtp_sender_->SetRtxSsrc(4321);
   rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
 
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
-                                       payload, sizeof(payload), 0));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr, nullptr));
 
   // Will send 2 full-size padding packets.
   rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
   rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
 
   StreamDataCounters rtp_stats;
   StreamDataCounters rtx_stats;
   rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
 
   // Payload + 1-byte generic header.
@@ skipping 76 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc