Index: media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc |
diff --git a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc b/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc |
deleted file mode 100644 |
index 16959e069e4f2e963d9483a4fad5bc8aec6db209..0000000000000000000000000000000000000000 |
--- a/media/cast/rtp_sender/rtp_packetizer/rtp_packetizer_unittest.cc |
+++ /dev/null |
@@ -1,154 +0,0 @@ |
-// 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 "base/memory/scoped_ptr.h" |
-#include "base/test/simple_test_tick_clock.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" |
-#include "testing/gmock/include/gmock/gmock.h" |
- |
-namespace media { |
-namespace cast { |
- |
-static const int kPayload = 127; |
-static const uint32 kTimestampMs = 10; |
-static const uint16 kSeqNum = 33; |
-static const int kMaxPacketLength = 1500; |
-static const int kSsrc = 0x12345; |
-static const unsigned int kFrameSize = 5000; |
-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), |
- expected_packet_id_(0), |
- expected_frame_id_(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) { |
- EXPECT_FALSE(rtp_header.is_key_frame); |
- EXPECT_EQ(expected_frame_id_, rtp_header.frame_id); |
- EXPECT_EQ(expected_packet_id_, rtp_header.packet_id); |
- EXPECT_EQ(expected_number_of_packets_ - 1, rtp_header.max_packet_id); |
- EXPECT_TRUE(rtp_header.is_reference); |
- EXPECT_EQ(expected_frame_id_ - 1u, rtp_header.reference_frame_id); |
- } |
- |
- virtual bool SendPackets(const PacketList& packets) OVERRIDE { |
- EXPECT_EQ(expected_number_of_packets_, static_cast<int>(packets.size())); |
- PacketList::const_iterator it = packets.begin(); |
- for (; it != packets.end(); ++it) { |
- ++packets_sent_; |
- RtpHeaderParser parser(it->data(), it->size()); |
- RtpCastHeader rtp_header; |
- parser.Parse(&rtp_header); |
- VerifyRtpHeader(rtp_header); |
- ++sequence_number_; |
- ++expected_packet_id_; |
- } |
- return true; |
- } |
- |
- virtual bool ResendPackets(const PacketList& 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_; |
- // Assuming packets arrive in sequence. |
- int expected_packet_id_; |
- uint32 expected_frame_id_; |
-}; |
- |
-class RtpPacketizerTest : public ::testing::Test { |
- protected: |
- RtpPacketizerTest() |
- :video_frame_(), |
- packet_storage_(&testing_clock_, 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_)); |
- } |
- |
- virtual ~RtpPacketizerTest() {} |
- |
- virtual void SetUp() { |
- video_frame_.key_frame = false; |
- video_frame_.frame_id = 0; |
- video_frame_.last_referenced_frame_id = kStartFrameId; |
- video_frame_.data.assign(kFrameSize, 123); |
- } |
- |
- base::SimpleTestTickClock testing_clock_; |
- 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. |
- int expected_num_of_packets = kFrameSize / kMaxPacketLength + 1; |
- transport_->SetExpectedNumberOfPackets(expected_num_of_packets); |
- |
- testing_clock_.Advance(base::TimeDelta::FromMilliseconds(kTimestampMs)); |
- rtp_packetizer_->IncomingEncodedVideoFrame(&video_frame_, |
- testing_clock_.NowTicks()); |
- EXPECT_EQ(expected_num_of_packets, rtp_packetizer_->send_packets_count()); |
- EXPECT_EQ(kFrameSize, rtp_packetizer_->send_octet_count()); |
-} |
- |
-} // namespace cast |
-} // namespace media |