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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc

Issue 2809613002: Revert of Implemented the GetSources() in native code. (Closed)
Patch Set: Created 3 years, 8 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
deleted file mode 100644
index 459e8e0315623841e30cfa29cedfe6f66202d048..0000000000000000000000000000000000000000
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
+++ /dev/null
@@ -1,260 +0,0 @@
-/*
- * Copyright (c) 2017 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 <memory>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
-#include "webrtc/test/gtest.h"
-
-namespace webrtc {
-
-const uint32_t kTestRate = 64000u;
-const uint8_t kTestPayload[] = {'t', 'e', 's', 't'};
-const uint8_t kPcmuPayloadType = 96;
-const int64_t kGetSourcesTimeoutMs = 10000;
-const int kSourceListsSize = 20;
-
-class RtpReceiverTest : public ::testing::Test {
- protected:
- RtpReceiverTest()
- : fake_clock_(123456),
- rtp_receiver_(
- RtpReceiver::CreateAudioReceiver(&fake_clock_,
- nullptr,
- nullptr,
- &rtp_payload_registry_)) {
- CodecInst voice_codec = {};
- voice_codec.pltype = kPcmuPayloadType;
- voice_codec.plfreq = 8000;
- voice_codec.rate = kTestRate;
- memcpy(voice_codec.plname, "PCMU", 5);
- rtp_receiver_->RegisterReceivePayload(voice_codec);
- }
- ~RtpReceiverTest() {}
-
- bool FindSourceByIdAndType(const std::vector<RtpSource>& sources,
- uint32_t source_id,
- RtpSourceType type,
- RtpSource* source) {
- for (size_t i = 0; i < sources.size(); ++i) {
- if (sources[i].source_id() == source_id &&
- sources[i].source_type() == type) {
- (*source) = sources[i];
- return true;
- }
- }
- return false;
- }
-
- SimulatedClock fake_clock_;
- RTPPayloadRegistry rtp_payload_registry_;
- std::unique_ptr<RtpReceiver> rtp_receiver_;
-};
-
-TEST_F(RtpReceiverTest, GetSources) {
- RTPHeader header;
- header.payloadType = kPcmuPayloadType;
- header.ssrc = 1;
- header.timestamp = fake_clock_.TimeInMilliseconds();
- header.numCSRCs = 2;
- header.arrOfCSRCs[0] = 111;
- header.arrOfCSRCs[1] = 222;
- PayloadUnion payload_specific = {AudioPayload()};
- bool in_order = false;
- RtpSource source(0, 0, RtpSourceType::SSRC);
-
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- auto sources = rtp_receiver_->GetSources();
- // One SSRC source and two CSRC sources.
- ASSERT_EQ(3u, sources.size());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, 222u, RtpSourceType::CSRC, &source));
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, 111u, RtpSourceType::CSRC, &source));
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
-
- // Advance the fake clock and the method is expected to return the
- // contributing source object with same source id and updated timestamp.
- fake_clock_.AdvanceTimeMilliseconds(1);
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- sources = rtp_receiver_->GetSources();
- ASSERT_EQ(3u, sources.size());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, 222u, RtpSourceType::CSRC, &source));
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, 111u, RtpSourceType::CSRC, &source));
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
-
- // Test the edge case that the sources are still there just before the
- // timeout.
- int64_t prev_timestamp = fake_clock_.TimeInMilliseconds();
- fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
- sources = rtp_receiver_->GetSources();
- ASSERT_EQ(3u, sources.size());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(prev_timestamp, source.timestamp_ms());
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, 222u, RtpSourceType::CSRC, &source));
- EXPECT_EQ(prev_timestamp, source.timestamp_ms());
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, 111u, RtpSourceType::CSRC, &source));
- EXPECT_EQ(prev_timestamp, source.timestamp_ms());
-
- // Time out.
- fake_clock_.AdvanceTimeMilliseconds(1);
- sources = rtp_receiver_->GetSources();
- // All the sources should be out of date.
- ASSERT_EQ(0u, sources.size());
-}
-
-// Test the case that the SSRC is changed.
-TEST_F(RtpReceiverTest, GetSourcesChangeSSRC) {
- int64_t prev_time = -1;
- int64_t cur_time = fake_clock_.TimeInMilliseconds();
- RTPHeader header;
- header.payloadType = kPcmuPayloadType;
- header.ssrc = 1;
- header.timestamp = cur_time;
- PayloadUnion payload_specific = {AudioPayload()};
- bool in_order = false;
- RtpSource source(0, 0, RtpSourceType::SSRC);
-
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- auto sources = rtp_receiver_->GetSources();
- ASSERT_EQ(1u, sources.size());
- EXPECT_EQ(1u, sources[0].source_id());
- EXPECT_EQ(cur_time, sources[0].timestamp_ms());
-
- // The SSRC is changed and the old SSRC is expected to be returned.
- fake_clock_.AdvanceTimeMilliseconds(100);
- prev_time = cur_time;
- cur_time = fake_clock_.TimeInMilliseconds();
- header.ssrc = 2;
- header.timestamp = cur_time;
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- sources = rtp_receiver_->GetSources();
- ASSERT_EQ(2u, sources.size());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 2u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(cur_time, source.timestamp_ms());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(prev_time, source.timestamp_ms());
-
- // The SSRC is changed again and happen to be changed back to 1. No
- // duplication is expected.
- fake_clock_.AdvanceTimeMilliseconds(100);
- header.ssrc = 1;
- header.timestamp = cur_time;
- prev_time = cur_time;
- cur_time = fake_clock_.TimeInMilliseconds();
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- sources = rtp_receiver_->GetSources();
- ASSERT_EQ(2u, sources.size());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(cur_time, source.timestamp_ms());
- ASSERT_TRUE(FindSourceByIdAndType(sources, 2u, RtpSourceType::SSRC, &source));
- EXPECT_EQ(prev_time, source.timestamp_ms());
-
- // Old SSRC source timeout.
- fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
- cur_time = fake_clock_.TimeInMilliseconds();
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- sources = rtp_receiver_->GetSources();
- ASSERT_EQ(1u, sources.size());
- EXPECT_EQ(1u, sources[0].source_id());
- EXPECT_EQ(cur_time, sources[0].timestamp_ms());
- EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
-}
-
-TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
- int64_t timestamp = fake_clock_.TimeInMilliseconds();
- bool in_order = false;
- RTPHeader header;
- header.payloadType = kPcmuPayloadType;
- header.timestamp = timestamp;
- PayloadUnion payload_specific = {AudioPayload()};
- header.numCSRCs = 1;
- RtpSource source(0, 0, RtpSourceType::SSRC);
-
- for (size_t i = 0; i < kSourceListsSize; ++i) {
- header.ssrc = i;
- header.arrOfCSRCs[0] = (i + 1);
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- }
-
- auto sources = rtp_receiver_->GetSources();
- // Expect |kSourceListsSize| SSRC sources and |kSourceListsSize| CSRC sources.
- ASSERT_TRUE(sources.size() == 2 * kSourceListsSize);
- for (size_t i = 0; i < kSourceListsSize; ++i) {
- // The SSRC source IDs are expected to be 19, 18, 17 ... 0
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
- EXPECT_EQ(timestamp, source.timestamp_ms());
-
- // The CSRC source IDs are expected to be 20, 19, 18 ... 1
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, (i + 1), RtpSourceType::CSRC, &source));
- EXPECT_EQ(timestamp, source.timestamp_ms());
- }
-
- fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
- for (size_t i = 0; i < kSourceListsSize; ++i) {
- // The SSRC source IDs are expected to be 19, 18, 17 ... 0
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
- EXPECT_EQ(timestamp, source.timestamp_ms());
-
- // The CSRC source IDs are expected to be 20, 19, 18 ... 1
- ASSERT_TRUE(
- FindSourceByIdAndType(sources, (i + 1), RtpSourceType::CSRC, &source));
- EXPECT_EQ(timestamp, source.timestamp_ms());
- }
-
- // Timeout. All the existing objects are out of date and are expected to be
- // removed.
- fake_clock_.AdvanceTimeMilliseconds(1);
- header.ssrc = 111;
- header.arrOfCSRCs[0] = 222;
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
- payload_specific, in_order));
- auto rtp_receiver_impl = static_cast<RtpReceiverImpl*>(rtp_receiver_.get());
- auto ssrc_sources = rtp_receiver_impl->ssrc_sources_for_testing();
- ASSERT_EQ(1u, ssrc_sources.size());
- EXPECT_EQ(111u, ssrc_sources.begin()->source_id());
- EXPECT_EQ(RtpSourceType::SSRC, ssrc_sources.begin()->source_type());
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
- ssrc_sources.begin()->timestamp_ms());
-
- auto csrc_sources = rtp_receiver_impl->csrc_sources_for_testing();
- ASSERT_EQ(1u, csrc_sources.size());
- EXPECT_EQ(222u, csrc_sources.begin()->source_id());
- EXPECT_EQ(RtpSourceType::CSRC, csrc_sources.begin()->source_type());
- EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
- csrc_sources.begin()->timestamp_ms());
-}
-
-} // namespace webrtc
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