Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 172 insertions, 0 deletions

diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
new file mode 100644
index 0000000000..eecb13757c
--- /dev/null
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2015 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 <list>
+
+#include "webrtc/base/checks.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/pacing/include/packet_router.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "webrtc/base/scoped_ptr.h"
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace webrtc {
+
+class PacketRouterTest : public ::testing::Test {
+ public:
+  PacketRouterTest() : packet_router_(new PacketRouter()) {}
+ protected:
+  const rtc::scoped_ptr<PacketRouter> packet_router_;
+};
+
+TEST_F(PacketRouterTest, TimeToSendPacket) {
+  MockRtpRtcp rtp_1;
+  MockRtpRtcp rtp_2;
+  packet_router_->AddRtpModule(&rtp_1);
+  packet_router_->AddRtpModule(&rtp_2);
+
+  const uint16_t kSsrc1 = 1234;
+  uint16_t sequence_number = 17;
+  uint64_t timestamp = 7890;
+  bool retransmission = false;
+
+  // Send on the first module by letting rtp_1 be sending with correct ssrc.
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
+  EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp,
+                                      retransmission))
+      .Times(1)
+      .WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
+                                               timestamp, retransmission));
+
+  // Send on the second module by letting rtp_2 be sending, but not rtp_1.
+  ++sequence_number;
+  timestamp += 30;
+  retransmission = true;
+  const uint16_t kSsrc2 = 4567;
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
+  EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, sequence_number, timestamp,
+                                      retransmission))
+      .Times(1)
+      .WillOnce(Return(true));
+  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc2, sequence_number,
+                                               timestamp, retransmission));
+
+  // No module is sending, hence no packet should be sent.
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
+                                               timestamp, retransmission));
+
+  // Add a packet with incorrect ssrc and test it's dropped in the router.
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
+  EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1 + kSsrc2, sequence_number,
+                                               timestamp, retransmission));
+
+  packet_router_->RemoveRtpModule(&rtp_1);
+
+  // rtp_1 has been removed, try sending a packet on that ssrc and make sure
+  // it is dropped as expected by not expecting any calls to rtp_1.
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
+  EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _)).Times(0);
+  EXPECT_TRUE(packet_router_->TimeToSendPacket(kSsrc1, sequence_number,
+                                               timestamp, retransmission));
+
+  packet_router_->RemoveRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, TimeToSendPadding) {
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+
+  MockRtpRtcp rtp_1;
+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
+  MockRtpRtcp rtp_2;
+  EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
+  packet_router_->AddRtpModule(&rtp_1);
+  packet_router_->AddRtpModule(&rtp_2);
+
+  // Default configuration, sending padding on all modules sending media,
+  // ordered by SSRC.
+  const size_t requested_padding_bytes = 1000;
+  const size_t sent_padding_bytes = 890;
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
+      .Times(1)
+      .WillOnce(Return(sent_padding_bytes));
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2,
+              TimeToSendPadding(requested_padding_bytes - sent_padding_bytes))
+      .Times(1)
+      .WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
+  EXPECT_EQ(requested_padding_bytes,
+            packet_router_->TimeToSendPadding(requested_padding_bytes));
+
+  // Let only the second module be sending and verify the padding request is
+  // routed there.
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, TimeToSendPadding(_))
+      .Times(1)
+      .WillOnce(Return(sent_padding_bytes));
+  EXPECT_EQ(sent_padding_bytes,
+            packet_router_->TimeToSendPadding(requested_padding_bytes));
+
+  // No sending module at all.
+  EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes)).Times(0);
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
+  EXPECT_CALL(rtp_2, TimeToSendPadding(_)).Times(0);
+  EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
+
+  packet_router_->RemoveRtpModule(&rtp_1);
+
+  // rtp_1 has been removed, try sending padding and make sure rtp_1 isn't asked
+  // to send by not expecting any calls. Instead verify rtp_2 is called.
+  EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
+  EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes)).Times(1);
+  EXPECT_EQ(0u, packet_router_->TimeToSendPadding(requested_padding_bytes));
+
+  packet_router_->RemoveRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, AllocateSequenceNumbers) {
+  const uint16_t kStartSeq = 0xFFF0;
+  const size_t kNumPackets = 32;
+
+  packet_router_->SetTransportWideSequenceNumber(kStartSeq - 1);
+
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    uint16_t seq = packet_router_->AllocateSequenceNumber();
+    uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
+    EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq);
+  }
+}
+}  // namespace webrtc