Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
This commit is contained in:
Mirko Bonadei
committed by
Commit Bot
Commit Bot
parent
6674846b4a
commit
bb547203bf
342
modules/rtp_rtcp/source/rtp_receiver_unittest.cc
Normal file
342
modules/rtp_rtcp/source/rtp_receiver_unittest.cc
Normal file
@ -0,0 +1,342 @@
|
||||
/*
|
||||
* 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/mocks/mock_rtp_rtcp.h"
|
||||
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
|
||||
#include "webrtc/test/gmock.h"
|
||||
#include "webrtc/test/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace {
|
||||
|
||||
using ::testing::NiceMock;
|
||||
using ::testing::UnorderedElementsAre;
|
||||
|
||||
const uint32_t kTestRate = 64000u;
|
||||
const uint8_t kTestPayload[] = {'t', 'e', 's', 't'};
|
||||
const uint8_t kPcmuPayloadType = 96;
|
||||
const int64_t kGetSourcesTimeoutMs = 10000;
|
||||
const uint32_t kSsrc1 = 123;
|
||||
const uint32_t kSsrc2 = 124;
|
||||
const uint32_t kCsrc1 = 111;
|
||||
const uint32_t kCsrc2 = 222;
|
||||
const bool kInOrder = true;
|
||||
|
||||
static uint32_t rtp_timestamp(int64_t time_ms) {
|
||||
return static_cast<uint32_t>(time_ms * kTestRate / 1000);
|
||||
}
|
||||
|
||||
} // namespace
|
||||
|
||||
class RtpReceiverTest : public ::testing::Test {
|
||||
protected:
|
||||
RtpReceiverTest()
|
||||
: fake_clock_(123456),
|
||||
rtp_receiver_(
|
||||
RtpReceiver::CreateAudioReceiver(&fake_clock_,
|
||||
&mock_rtp_data_,
|
||||
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_;
|
||||
NiceMock<MockRtpData> mock_rtp_data_;
|
||||
RTPPayloadRegistry rtp_payload_registry_;
|
||||
std::unique_ptr<RtpReceiver> rtp_receiver_;
|
||||
};
|
||||
|
||||
TEST_F(RtpReceiverTest, GetSources) {
|
||||
int64_t now_ms = fake_clock_.TimeInMilliseconds();
|
||||
|
||||
RTPHeader header;
|
||||
header.payloadType = kPcmuPayloadType;
|
||||
header.ssrc = kSsrc1;
|
||||
header.timestamp = rtp_timestamp(now_ms);
|
||||
header.numCSRCs = 2;
|
||||
header.arrOfCSRCs[0] = kCsrc1;
|
||||
header.arrOfCSRCs[1] = kCsrc2;
|
||||
PayloadUnion payload_specific = {AudioPayload()};
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
auto sources = rtp_receiver_->GetSources();
|
||||
// One SSRC source and two CSRC sources.
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC),
|
||||
RtpSource(now_ms, kCsrc1, RtpSourceType::CSRC),
|
||||
RtpSource(now_ms, kCsrc2, RtpSourceType::CSRC)));
|
||||
|
||||
// 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, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
now_ms = fake_clock_.TimeInMilliseconds();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC),
|
||||
RtpSource(now_ms, kCsrc1, RtpSourceType::CSRC),
|
||||
RtpSource(now_ms, kCsrc2, RtpSourceType::CSRC)));
|
||||
|
||||
// Test the edge case that the sources are still there just before the
|
||||
// timeout.
|
||||
int64_t prev_time_ms = fake_clock_.TimeInMilliseconds();
|
||||
fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources,
|
||||
UnorderedElementsAre(
|
||||
RtpSource(prev_time_ms, kSsrc1, RtpSourceType::SSRC),
|
||||
RtpSource(prev_time_ms, kCsrc1, RtpSourceType::CSRC),
|
||||
RtpSource(prev_time_ms, kCsrc2, RtpSourceType::CSRC)));
|
||||
|
||||
// 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_ms = -1;
|
||||
int64_t now_ms = fake_clock_.TimeInMilliseconds();
|
||||
|
||||
RTPHeader header;
|
||||
header.payloadType = kPcmuPayloadType;
|
||||
header.ssrc = kSsrc1;
|
||||
header.timestamp = rtp_timestamp(now_ms);
|
||||
PayloadUnion payload_specific = {AudioPayload()};
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
auto sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
|
||||
|
||||
// The SSRC is changed and the old SSRC is expected to be returned.
|
||||
fake_clock_.AdvanceTimeMilliseconds(100);
|
||||
prev_time_ms = now_ms;
|
||||
now_ms = fake_clock_.TimeInMilliseconds();
|
||||
header.ssrc = kSsrc2;
|
||||
header.timestamp = rtp_timestamp(now_ms);
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(prev_time_ms, kSsrc1, RtpSourceType::SSRC),
|
||||
RtpSource(now_ms, kSsrc2, RtpSourceType::SSRC)));
|
||||
|
||||
// The SSRC is changed again and happen to be changed back to 1. No
|
||||
// duplication is expected.
|
||||
fake_clock_.AdvanceTimeMilliseconds(100);
|
||||
header.ssrc = kSsrc1;
|
||||
header.timestamp = rtp_timestamp(now_ms);
|
||||
prev_time_ms = now_ms;
|
||||
now_ms = fake_clock_.TimeInMilliseconds();
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(prev_time_ms, kSsrc2, RtpSourceType::SSRC),
|
||||
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
|
||||
|
||||
// Old SSRC source timeout.
|
||||
fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
|
||||
now_ms = fake_clock_.TimeInMilliseconds();
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
|
||||
}
|
||||
|
||||
TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
|
||||
int64_t now_ms = fake_clock_.TimeInMilliseconds();
|
||||
|
||||
RTPHeader header;
|
||||
header.payloadType = kPcmuPayloadType;
|
||||
header.timestamp = rtp_timestamp(now_ms);
|
||||
PayloadUnion payload_specific = {AudioPayload()};
|
||||
header.numCSRCs = 1;
|
||||
size_t kSourceListSize = 20;
|
||||
|
||||
for (size_t i = 0; i < kSourceListSize; ++i) {
|
||||
header.ssrc = i;
|
||||
header.arrOfCSRCs[0] = (i + 1);
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload,
|
||||
sizeof(kTestPayload),
|
||||
payload_specific, !kInOrder));
|
||||
}
|
||||
|
||||
RtpSource source(0, 0, RtpSourceType::SSRC);
|
||||
auto sources = rtp_receiver_->GetSources();
|
||||
// Expect |kSourceListSize| SSRC sources and |kSourceListSize| CSRC sources.
|
||||
ASSERT_EQ(2 * kSourceListSize, sources.size());
|
||||
for (size_t i = 0; i < kSourceListSize; ++i) {
|
||||
// The SSRC source IDs are expected to be 19, 18, 17 ... 0
|
||||
ASSERT_TRUE(
|
||||
FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
|
||||
EXPECT_EQ(now_ms, 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(now_ms, source.timestamp_ms());
|
||||
}
|
||||
|
||||
fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
|
||||
for (size_t i = 0; i < kSourceListSize; ++i) {
|
||||
// The SSRC source IDs are expected to be 19, 18, 17 ... 0
|
||||
ASSERT_TRUE(
|
||||
FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
|
||||
EXPECT_EQ(now_ms, 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(now_ms, source.timestamp_ms());
|
||||
}
|
||||
|
||||
// Timeout. All the existing objects are out of date and are expected to be
|
||||
// removed.
|
||||
fake_clock_.AdvanceTimeMilliseconds(1);
|
||||
header.ssrc = kSsrc1;
|
||||
header.arrOfCSRCs[0] = kCsrc1;
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
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(kSsrc1, 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(kCsrc1, csrc_sources.begin()->source_id());
|
||||
EXPECT_EQ(RtpSourceType::CSRC, csrc_sources.begin()->source_type());
|
||||
EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
|
||||
csrc_sources.begin()->timestamp_ms());
|
||||
}
|
||||
|
||||
// The audio level from the RTPHeader extension should be stored in the
|
||||
// RtpSource with the matching SSRC.
|
||||
TEST_F(RtpReceiverTest, GetSourcesContainsAudioLevelExtension) {
|
||||
RTPHeader header;
|
||||
int64_t time1_ms = fake_clock_.TimeInMilliseconds();
|
||||
header.payloadType = kPcmuPayloadType;
|
||||
header.ssrc = kSsrc1;
|
||||
header.timestamp = rtp_timestamp(time1_ms);
|
||||
header.extension.hasAudioLevel = true;
|
||||
header.extension.audioLevel = 10;
|
||||
PayloadUnion payload_specific = {AudioPayload()};
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
auto sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
|
||||
time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
|
||||
|
||||
// Receive a packet from a different SSRC with a different level and check
|
||||
// that they are both remembered.
|
||||
fake_clock_.AdvanceTimeMilliseconds(1);
|
||||
int64_t time2_ms = fake_clock_.TimeInMilliseconds();
|
||||
header.ssrc = kSsrc2;
|
||||
header.timestamp = rtp_timestamp(time2_ms);
|
||||
header.extension.hasAudioLevel = true;
|
||||
header.extension.audioLevel = 20;
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources,
|
||||
UnorderedElementsAre(
|
||||
RtpSource(time1_ms, kSsrc1, RtpSourceType::SSRC, 10),
|
||||
RtpSource(time2_ms, kSsrc2, RtpSourceType::SSRC, 20)));
|
||||
|
||||
// Receive a packet from the first SSRC again and check that the level is
|
||||
// updated.
|
||||
fake_clock_.AdvanceTimeMilliseconds(1);
|
||||
int64_t time3_ms = fake_clock_.TimeInMilliseconds();
|
||||
header.ssrc = kSsrc1;
|
||||
header.timestamp = rtp_timestamp(time3_ms);
|
||||
header.extension.hasAudioLevel = true;
|
||||
header.extension.audioLevel = 30;
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources,
|
||||
UnorderedElementsAre(
|
||||
RtpSource(time3_ms, kSsrc1, RtpSourceType::SSRC, 30),
|
||||
RtpSource(time2_ms, kSsrc2, RtpSourceType::SSRC, 20)));
|
||||
}
|
||||
|
||||
TEST_F(RtpReceiverTest,
|
||||
MissingAudioLevelHeaderExtensionClearsRtpSourceAudioLevel) {
|
||||
RTPHeader header;
|
||||
int64_t time1_ms = fake_clock_.TimeInMilliseconds();
|
||||
header.payloadType = kPcmuPayloadType;
|
||||
header.ssrc = kSsrc1;
|
||||
header.timestamp = rtp_timestamp(time1_ms);
|
||||
header.extension.hasAudioLevel = true;
|
||||
header.extension.audioLevel = 10;
|
||||
PayloadUnion payload_specific = {AudioPayload()};
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
auto sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
|
||||
time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
|
||||
|
||||
// Receive a second packet without the audio level header extension and check
|
||||
// that the audio level is cleared.
|
||||
fake_clock_.AdvanceTimeMilliseconds(1);
|
||||
int64_t time2_ms = fake_clock_.TimeInMilliseconds();
|
||||
header.timestamp = rtp_timestamp(time2_ms);
|
||||
header.extension.hasAudioLevel = false;
|
||||
|
||||
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
||||
header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
|
||||
sources = rtp_receiver_->GetSources();
|
||||
EXPECT_THAT(sources, UnorderedElementsAre(
|
||||
RtpSource(time2_ms, kSsrc1, RtpSourceType::SSRC)));
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
Reference in New Issue
Block a user