Unify FrameType and VideoFrameType.

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 1248 matching lines @@
 
   EXPECT_EQ(0, memcmp(extension, payload_data - sizeof(extension),
                       sizeof(extension)));
 }
 
 // As RFC4733, named telephone events are carried as part of the audio stream
 // and must use the same sequence number and timestamp base as the regular
 // audio channel.
 // This test checks the marker bit for the first packet and the consequent
 // packets of the same telephone event. Since it is specifically for DTMF
-// events, ignoring audio packets and sending kFrameEmpty instead of those.
+// events, ignoring audio packets and sending kEmptyFrame instead of those.
 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
   uint8_t payload_type = 126;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 0,
                                             0, 0));
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
   strcpy(payload_name, "payload_name");
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, 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(kFrameEmpty, payload_type,
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms, 0, nullptr, 0,
                                              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(kFrameEmpty, payload_type,
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 2000, 0, nullptr,
                                              0, nullptr));
   rtc::scoped_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(kFrameEmpty, payload_type,
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 4000, 0, nullptr,
                                              0, 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) {
@@ skipping 103 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