1 files changed, 145 insertions, 0 deletions
diff --git a/chromium/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc b/chromium/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc
new file mode 100644
index 00000000000..bed7cba2e8f
--- /dev/null
+++ b/chromium/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc
@@ -0,0 +1,145 @@
+// Copyright 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "media/cast/rtp_sender/rtp_packetizer/rtp_packetizer.h"
+
+#include <gtest/gtest.h>
+
+#include "base/memory/scoped_ptr.h"
+#include "media/cast/cast_config.h"
+#include "media/cast/pacing/paced_sender.h"
+#include "media/cast/rtp_common/rtp_defines.h"
+#include "media/cast/rtp_sender/packet_storage/packet_storage.h"
+#include "media/cast/rtp_sender/rtp_packetizer/test/rtp_header_parser.h"
+
+namespace media {
+namespace cast {
+
+static const int kPayload = 127;
+static const uint32 kTimestampMs = 10;
+static const uint16 kSeqNum = 33;
+static const int kTimeOffset = 22222;
+static const int kMaxPacketLength = 1500;
+static const bool kMarkerBit = true;
+static const int kSsrc = 0x12345;
+static const uint8 kFrameId = 1;
+static const unsigned int kFrameSize = 5000;
+static const int kTotalHeaderLength = 19;
+static const int kMaxPacketStorageTimeMs = 300;
+
+class TestRtpPacketTransport : public PacedPacketSender {
+ public:
+  explicit TestRtpPacketTransport(RtpPacketizerConfig config)
+       : config_(config),
+         sequence_number_(kSeqNum),
+         packets_sent_(0),
+         expected_number_of_packets_(0) {}
+
+  void VerifyRtpHeader(const RtpCastHeader& rtp_header) {
+    VerifyCommonRtpHeader(rtp_header);
+    VerifyCastRtpHeader(rtp_header);
+  }
+
+  void VerifyCommonRtpHeader(const RtpCastHeader& rtp_header) {
+    EXPECT_EQ(expected_number_of_packets_ == packets_sent_,
+        rtp_header.webrtc.header.markerBit);
+    EXPECT_EQ(kPayload, rtp_header.webrtc.header.payloadType);
+    EXPECT_EQ(sequence_number_, rtp_header.webrtc.header.sequenceNumber);
+    EXPECT_EQ(kTimestampMs * 90, rtp_header.webrtc.header.timestamp);
+    EXPECT_EQ(config_.ssrc, rtp_header.webrtc.header.ssrc);
+    EXPECT_EQ(0, rtp_header.webrtc.header.numCSRCs);
+  }
+
+  void VerifyCastRtpHeader(const RtpCastHeader& rtp_header) {
+    // TODO(mikhal)
+  }
+
+  virtual bool SendPacket(const std::vector<uint8>& packet,
+                          int num_packets) OVERRIDE {
+    EXPECT_EQ(expected_number_of_packets_, num_packets);
+    ++packets_sent_;
+    RtpHeaderParser parser(packet.data(), packet.size());
+    RtpCastHeader rtp_header;
+    parser.Parse(&rtp_header);
+    VerifyRtpHeader(rtp_header);
+    ++sequence_number_;
+    return true;
+  }
+
+  virtual bool ResendPacket(const std::vector<uint8>& packet,
+                            int num_of_packets) OVERRIDE {
+    EXPECT_TRUE(false);
+    return false;
+  }
+
+  virtual bool SendRtcpPacket(const std::vector<uint8>& packet) OVERRIDE {
+    EXPECT_TRUE(false);
+    return false;
+  }
+
+  void SetExpectedNumberOfPackets(int num) {
+    expected_number_of_packets_ = num;
+  }
+
+  RtpPacketizerConfig config_;
+  uint32 sequence_number_;
+  int packets_sent_;
+  int expected_number_of_packets_;
+};
+
+class RtpPacketizerTest : public ::testing::Test {
+ protected:
+  RtpPacketizerTest()
+      :video_frame_(),
+       packet_storage_(kMaxPacketStorageTimeMs) {
+    config_.sequence_number = kSeqNum;
+    config_.ssrc = kSsrc;
+    config_.payload_type = kPayload;
+    config_.max_payload_length = kMaxPacketLength;
+    transport_.reset(new TestRtpPacketTransport(config_));
+    rtp_packetizer_.reset(
+        new RtpPacketizer(transport_.get(), &packet_storage_, config_));
+  }
+
+  ~RtpPacketizerTest() {}
+
+  void SetUp() {
+    video_frame_.key_frame = false;
+    video_frame_.frame_id = kFrameId;
+    video_frame_.last_referenced_frame_id = kFrameId - 1;
+    video_frame_.data.assign(kFrameSize, 123);
+  }
+
+  scoped_ptr<RtpPacketizer> rtp_packetizer_;
+  RtpPacketizerConfig config_;
+  scoped_ptr<TestRtpPacketTransport> transport_;
+  EncodedVideoFrame video_frame_;
+  PacketStorage packet_storage_;
+};
+
+TEST_F(RtpPacketizerTest, SendStandardPackets) {
+  int expected_num_of_packets = kFrameSize / kMaxPacketLength + 1;
+  transport_->SetExpectedNumberOfPackets(expected_num_of_packets);
+
+  base::TimeTicks time;
+  time += base::TimeDelta::FromMilliseconds(kTimestampMs);
+    rtp_packetizer_->IncomingEncodedVideoFrame(&video_frame_,time);
+}
+
+TEST_F(RtpPacketizerTest, Stats) {
+  EXPECT_FALSE(rtp_packetizer_->send_packets_count());
+  EXPECT_FALSE(rtp_packetizer_->send_octet_count());
+  // Insert packets at varying lengths.
+  unsigned int expected_num_of_packets = kFrameSize / kMaxPacketLength + 1;
+  transport_->SetExpectedNumberOfPackets(expected_num_of_packets);
+
+  base::TimeTicks time;
+  time += base::TimeDelta::FromMilliseconds(kTimestampMs);
+  rtp_packetizer_->IncomingEncodedVideoFrame(&video_frame_, time);
+  EXPECT_EQ(expected_num_of_packets, rtp_packetizer_->send_packets_count());
+  EXPECT_EQ(kFrameSize, rtp_packetizer_->send_octet_count());
+}
+
+}  // namespace cast
+}  // namespace media