RtpPacket::payload() return rtc::ArrayView instead of raw pointer

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 860 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};
 
   // Send keyframe
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
                                             4321, payload, sizeof(payload),
                                             nullptr, nullptr, nullptr));
 
-  const uint8_t* payload_data = transport_.last_sent_packet().payload();
-  uint8_t generic_header = *payload_data++;
-
-  ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
-            transport_.last_sent_packet().payload_size());
-
+  auto sent_payload = transport_.last_sent_packet().payload();
+  uint8_t generic_header = sent_payload[0];
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
-
-  EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
+  EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));

[brandtr, 2016/11/21 07:49:22]

Nice!

 
   // Send delta frame
   payload[0] = 13;
   payload[1] = 42;
   payload[4] = 13;
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
       nullptr, nullptr, nullptr));
 
-  payload_data = transport_.last_sent_packet().payload();
-  generic_header = *payload_data++;
-
+  sent_payload = transport_.last_sent_packet().payload();
+  generic_header = sent_payload[0];
   EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
-
-  ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
-            transport_.last_sent_packet().payload_size());
-
-  EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
+  EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
 }
 
 TEST_F(RtpSenderTest, SendFlexfecPackets) {
   constexpr int kMediaPayloadType = 127;
   constexpr int kFlexfecPayloadType = 118;
   constexpr uint32_t kMediaSsrc = 1234;
   constexpr uint32_t kFlexfecSsrc = 5678;
   const std::vector<RtpExtension> kNoRtpExtensions;
   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
                                kNoRtpExtensions, &fake_clock_);
@@ skipping 330 matching lines @@
   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_TRUE(rtp_sender_->SendOutgoingData(
                       kAudioFrameCN, payload_type, 1234, 4321, payload,
                       sizeof(payload), nullptr, nullptr, nullptr));
 
-  const uint8_t* payload_data = transport_.last_sent_packet().payload();
-
-  ASSERT_EQ(sizeof(payload), transport_.last_sent_packet().payload_size());
-  EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
+  auto sent_payload = transport_.last_sent_packet().payload();
+  EXPECT_THAT(sent_payload, ElementsAreArray(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_TRUE(rtp_sender_->SendOutgoingData(
                       kAudioFrameCN, payload_type, 1234, 4321, payload,
                       sizeof(payload), nullptr, nullptr, nullptr));
 
-  const uint8_t* payload_data = transport_.last_sent_packet().payload();
-
-  ASSERT_EQ(sizeof(payload), transport_.last_sent_packet().payload_size());
-  EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
+  auto sent_payload = transport_.last_sent_packet().payload();
+  EXPECT_THAT(sent_payload, ElementsAreArray(payload));
   // Verify AudioLevel extension.
   bool voice_activity;
   uint8_t audio_level;
   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
       &voice_activity, &audio_level));
   EXPECT_EQ(kAudioLevel, audio_level);
   EXPECT_FALSE(voice_activity);
 }
 
 // As RFC4733, named telephone events are carried as part of the audio stream
@@ skipping 242 matching lines @@
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
   rtp_sender_->SetTransportOverhead(28);
   rtp_sender_->SetTransportOverhead(28);
 
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
   SendGenericPayload();
   SendGenericPayload();
 }
 
 }  // namespace webrtc