RtpPacket class introduced.

webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc

+/*
+ *  Copyright (c) 2016 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.
+ */
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+
+using testing::ElementsAreArray;
+using testing::make_tuple;
+
+namespace webrtc {
+namespace {
+const int8_t kPayloadType = 100;
+const uint32_t kSsrc = 0x12345678;
+const uint16_t kSeqNum = 88;
+const uint32_t kTimestamp = 0x65431278;
+const uint8_t kTransmissionOffsetExtensionId = 1;
+const uint8_t kAudioLevelExtensionId = 9;
+const int32_t kTimeOffset = 0x56ce;
+const bool kVoiceActive = true;
+const uint8_t kAudioLevel = 0x5a;
+// const size_t kMaxPaddingSize = 224u;

[sprang_webrtc, 2016/04/20 07:49:35]

Unused?

[danilchap, 2016/04/20 08:52:54]

Oops, made used.

+const uint8_t kMinimumPacket[] = {0x80, kPayloadType, 0x00, kSeqNum,
+                                  0x65, 0x43, 0x12, 0x78,
+                                  0x12, 0x34, 0x56, 0x78};
+const uint8_t kPacketWithTO[] = {
+    0x90, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x01,
+    0x12, 0x00, 0x56, 0xce};
+
+const uint8_t kPacketWithTOAndAL[] = {
+    0x90, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0xbe, 0xde, 0x00, 0x02,
+    0x12, 0x00, 0x56, 0xce,
+    0x90, 0x80|kAudioLevel, 0x00, 0x00};
+
+const uint32_t kCsrcs[] = {0x34567890, 0x32435465};
+const uint8_t kPayload[] = {'p', 'a', 'y', 'l', 'o', 'a', 'd'};
+const uint8_t kPacketPaddingSize = 8;
+const uint8_t kPacket[] = {
+    0xb2, kPayloadType, 0x00, kSeqNum,
+    0x65, 0x43, 0x12, 0x78,
+    0x12, 0x34, 0x56, 0x78,
+    0x34, 0x56, 0x78, 0x90,
+    0x32, 0x43, 0x54, 0x65,
+    0xbe, 0xde, 0x00, 0x01,
+    0x12, 0x00, 0x56, 0xce,
+    'p', 'a', 'y', 'l', 'o', 'a', 'd',
+    'p', 'a', 'd', 'd', 'i', 'n', 'g', kPacketPaddingSize};
+
+}  // namespace
+
+TEST(RtpPacketTest, CreateMinimum) {
+  RtpPacketToSend packet(nullptr);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  EXPECT_THAT(kMinimumPacket, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWithExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  EXPECT_THAT(kPacketWithTO, ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, CreateWith2Extensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.SetExtension<TransmissionOffset>(kTimeOffset);
+  packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel);
+  EXPECT_THAT(kPacketWithTOAndAL,
+              ElementsAreArray(packet.data(), packet.size()));
+}
+
+TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
+  const size_t kPayloadSize = 4;
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  RtpPacketToSend packet(&extensions);
+
+  EXPECT_TRUE(packet.ReserveExtension<TransmissionOffset>());
+  packet.AllocatePayload(kPayloadSize);
+  // Can't set extension after payload.
+  EXPECT_FALSE(packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel));
+  // Unless reserved.
+  EXPECT_TRUE(packet.SetExtension<TransmissionOffset>(kTimeOffset));
+}
+
+TEST(RtpPacketTest, CreatePurePadding) {
+  const uint8_t kPaddingSize = 0xfe;
+  RtpPacketToSend packet(nullptr, 12 + kPaddingSize);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  Random random(0x123456789);
+
+  EXPECT_LT(packet.size(), packet.capacity());
+  EXPECT_FALSE(packet.SetPadding(kPaddingSize + 1, &random));
+  EXPECT_TRUE(packet.SetPadding(kPaddingSize, &random));
+  EXPECT_EQ(packet.size(), packet.capacity());
+}
+
+TEST(RtpPacketTest, CreateUnalignedPadding) {
+  const size_t kPayloadSize = 3;  // Make padding start at unaligned address.
+  const uint8_t kPaddingSize = 0xfe;
+  RtpPacketToSend packet(nullptr, 12 + kPayloadSize + kPaddingSize);
+  packet.SetPayloadType(kPayloadType);
+  packet.SetSequenceNumber(kSeqNum);
+  packet.SetTimestamp(kTimestamp);
+  packet.SetSsrc(kSsrc);
+  packet.AllocatePayload(kPayloadSize);
+  Random r(0x123456789);
+
+  EXPECT_LT(packet.size(), packet.capacity());
+  EXPECT_TRUE(packet.SetPadding(kPaddingSize, &r));
+  EXPECT_EQ(packet.size(), packet.capacity());
+}
+
+TEST(RtpPacketTest, ParseMinimum) {
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(kMinimumPacket, sizeof(kMinimumPacket)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_EQ(0u, packet.padding_size());
+  EXPECT_EQ(0u, packet.payload_size());
+}
+
+TEST(RtpPacketTest, ParseBuffer) {
+  rtc::Buffer unparsed(kMinimumPacket);
+  const uint8_t* raw = unparsed.data();
+
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(std::move(unparsed)));
+  EXPECT_EQ(raw, packet.data());  // Expect packet took over the buffer.
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_EQ(0u, packet.padding_size());
+  EXPECT_EQ(0u, packet.payload_size());
+}
+
+TEST(RtpPacketTest, ParseWithExtension) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  EXPECT_EQ(0u, packet.payload_size());
+  EXPECT_EQ(0u, packet.padding_size());
+}
+
+TEST(RtpPacketTest, ParseWith2Extensions) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacketWithTOAndAL, sizeof(kPacketWithTOAndAL)));
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  bool voice_active;
+  uint8_t audio_level;
+  EXPECT_TRUE(packet.GetExtension<AudioLevel>(&voice_active, &audio_level));
+  EXPECT_EQ(kVoiceActive, voice_active);
+  EXPECT_EQ(kAudioLevel, audio_level);
+}
+
+TEST(RtpPacketTest, ParseWithAllFeatures) {
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+  RtpPacketReceived packet(&extensions);
+  EXPECT_TRUE(packet.Parse(kPacket, sizeof(kPacket)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+  EXPECT_THAT(packet.Csrcs(), ElementsAreArray(kCsrcs));
+  EXPECT_THAT(make_tuple(packet.payload(), packet.payload_size()),
+              ElementsAreArray(kPayload));
+  EXPECT_EQ(kPacketPaddingSize, packet.padding_size());
+  int32_t time_offset;
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+}
+
+TEST(RtpPacketTest, ParseWithExtensionDelayed) {
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
+  EXPECT_EQ(kPayloadType, packet.PayloadType());
+  EXPECT_EQ(kSeqNum, packet.SequenceNumber());
+  EXPECT_EQ(kTimestamp, packet.Timestamp());
+  EXPECT_EQ(kSsrc, packet.Ssrc());
+
+  RtpPacketToSend::ExtensionManager extensions;
+  extensions.Register(kRtpExtensionTransmissionTimeOffset,
+                      kTransmissionOffsetExtensionId);
+
+  int32_t time_offset;
+  EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  packet.IdentifyExtensions(&extensions);
+  EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
+  EXPECT_EQ(kTimeOffset, time_offset);
+  EXPECT_EQ(0u, packet.payload_size());
+  EXPECT_EQ(0u, packet.padding_size());
+}
+
+}  // namespace webrtc