zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Move RtpSenderAudioTest to its own file

Browse Source 
Update RtpSenderAudioTest to call methods on RTPSenderAudio rather
than RTPSender, when possible. In particular, avoid
RTPSender::SendOutgoingData. Drop parameterization on the
WebRTC-SendSideBwe-WithOverhead field trial, since that appears
unrelated to these tests.

Also delete some unused parts of the RtpSender test.

Bug: webrtc:7135
Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0
Reviewed-on: https://webrtc-review.googlesource.com/c/120920
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26516}
This commit is contained in:
Niels Möller
2019-02-01 14:13:29 +01:00
committed by Commit Bot
parent 05cf6be726
commit a34d7766c5
3 changed files with 175 additions and 131 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
modules/rtp_rtcp/BUILD.gn
View File

@ -411,6 +411,7 @@ if (rtc_include_tests) {
"source/rtp_packet_history_unittest.cc",
"source/rtp_packet_unittest.cc",
"source/rtp_rtcp_impl_unittest.cc",
"source/rtp_sender_audio_unittest.cc",
"source/rtp_sender_unittest.cc",
"source/rtp_utility_unittest.cc",
"source/time_util_unittest.cc",

173
modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc Normal file
View File

@ -0,0 +1,173 @@
/*
* Copyright (c) 2019 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 <vector>
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
const int kAudioLevelExtensionId = 9;
const uint16_t kSeqNum = 33;
const uint32_t kSsrc = 725242;
const uint8_t kAudioLevel = 0x5a;
const uint64_t kStartTime = 123456789;
using ::testing::_;
using ::testing::ElementsAreArray;
class LoopbackTransportTest : public webrtc::Transport {
public:
LoopbackTransportTest() {
receivers_extensions_.Register(kRtpExtensionAudioLevel,
kAudioLevelExtensionId);
}
bool SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& /*options*/) override {
sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
EXPECT_TRUE(sent_packets_.back().Parse(data, len));
return true;
}
bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); }
int packets_sent() { return sent_packets_.size(); }
private:
RtpHeaderExtensionMap receivers_extensions_;
std::vector<RtpPacketReceived> sent_packets_;
};
} // namespace
class RtpSenderAudioTest : public ::testing::Test {
public:
RtpSenderAudioTest()
: fake_clock_(kStartTime),
rtp_sender_(true,
&fake_clock_,
&transport_,
nullptr,
nullptr,
nullptr,
nullptr,
nullptr,
nullptr,
nullptr,
nullptr,
nullptr,
nullptr,
false,
nullptr,
false,
false),
rtp_sender_audio_(&fake_clock_, &rtp_sender_) {
rtp_sender_.SetSSRC(kSsrc);
rtp_sender_.SetSequenceNumber(kSeqNum);
}
SimulatedClock fake_clock_;
LoopbackTransportTest transport_;
RTPSender rtp_sender_;
RTPSenderAudio rtp_sender_audio_;
};
TEST_F(RtpSenderAudioTest, SendAudio) {
const char payload_name[] = "PAYLOAD_NAME";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload(
payload_name, payload_type, 48000, 0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
ASSERT_TRUE(rtp_sender_audio_.SendAudio(kAudioFrameCN, payload_type, 4321,
payload, sizeof(payload)));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
}
TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_audio_.SetAudioLevel(kAudioLevel));
EXPECT_EQ(0, rtp_sender_.RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
const char payload_name[] = "PAYLOAD_NAME";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload(
payload_name, payload_type, 48000, 0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
ASSERT_TRUE(rtp_sender_audio_.SendAudio(kAudioFrameCN, payload_type, 4321,
payload, sizeof(payload)));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
// Verify AudioLevel extension.
bool voice_activity;
uint8_t audio_level;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
&voice_activity, &audio_level));
EXPECT_EQ(kAudioLevel, audio_level);
EXPECT_FALSE(voice_activity);
}
// As RFC4733, named telephone events are carried as part of the audio stream
// and must use the same sequence number and timestamp base as the regular
// audio channel.
// This test checks the marker bit for the first packet and the consequent
// packets of the same telephone event. Since it is specifically for DTMF
// events, ignoring audio packets and sending kEmptyFrame instead of those.
TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
const char* kDtmfPayloadName = "telephone-event";
const uint32_t kPayloadFrequency = 8000;
const uint8_t kPayloadType = 126;
ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload(
kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0));
// For Telephone events, payload is not added to the registered payload list,
// it will register only the payload used for audio stream.
// Registering the payload again for audio stream with different payload name.
const char* kPayloadName = "payload_name";
ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload(
kPayloadName, kPayloadType, kPayloadFrequency, 1, 0));
// Start time is arbitrary.
uint32_t capture_timestamp = fake_clock_.TimeInMilliseconds();
// DTMF event key=9, duration=500 and attenuationdB=10
rtp_sender_audio_.SendTelephoneEvent(9, 500, 10);
// During start, it takes the starting timestamp as last sent timestamp.
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
ASSERT_TRUE(rtp_sender_audio_.SendAudio(kEmptyFrame, kPayloadType,
capture_timestamp, nullptr, 0));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
ASSERT_TRUE(rtp_sender_audio_.SendAudio(
kEmptyFrame, kPayloadType, capture_timestamp + 2000, nullptr, 0));
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(transport_.last_sent_packet().Marker());
ASSERT_TRUE(rtp_sender_audio_.SendAudio(
kEmptyFrame, kPayloadType, capture_timestamp + 4000, nullptr, 0));
// Marker Bit should be set to 0 for rest of the packets.
EXPECT_FALSE(transport_.last_sent_packet().Marker());
}
} // namespace webrtc

132
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -31,7 +31,6 @@
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/buffer.h"
#include "rtc_base/rate_limiter.h"
#include "test/field_trial.h"
#include "test/gmock.h"
@ -57,9 +56,7 @@ const uint32_t kTimestamp = 10;
const uint16_t kSeqNum = 33;
const uint32_t kSsrc = 725242;
const int kMaxPacketLength = 1500;
const uint8_t kAudioLevel = 0x5a;
const uint16_t kTransportSequenceNumber = 0xaabbu;
const int kAudioPayload = 103;
const uint64_t kStartTime = 123456789;
const size_t kMaxPaddingSize = 224u;
const size_t kGenericHeaderLength = 1;
@ -179,7 +176,6 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
mock_paced_sender_(),
retransmission_rate_limiter_(&fake_clock_, 1000),
rtp_sender_(),
payload_(kPayload),
transport_(),
kMarkerBit(true),
field_trials_(GetParam() ? "WebRTC-SendSideBwe-WithOverhead/Enabled/"
@ -207,31 +203,10 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
testing::StrictMock<MockTransportFeedbackObserver> feedback_observer_;
RateLimiter retransmission_rate_limiter_;
std::unique_ptr<RTPSender> rtp_sender_;
int payload_;
LoopbackTransportTest transport_;
const bool kMarkerBit;
test::ScopedFieldTrials field_trials_;
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
}
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) {
VerifyRTPHeaderCommon(rtp_header, marker_bit, 0);
}
void VerifyRTPHeaderCommon(const RTPHeader& rtp_header,
bool marker_bit,
uint8_t number_of_csrcs) {
EXPECT_EQ(marker_bit, rtp_header.markerBit);
EXPECT_EQ(payload_, rtp_header.payloadType);
EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
EXPECT_EQ(kTimestamp, rtp_header.timestamp);
EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc);
EXPECT_EQ(number_of_csrcs, rtp_header.numCSRCs);
EXPECT_EQ(0U, rtp_header.paddingLength);
}
std::unique_ptr<RtpPacketToSend> BuildRtpPacket(int payload_type,
bool marker_bit,
uint32_t timestamp,
@ -1549,21 +1524,6 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
rtp_sender_.reset();
}
class RtpSenderAudioTest : public RtpSenderTest {
protected:
RtpSenderAudioTest() {}
void SetUp() override {
payload_ = kAudioPayload;
rtp_sender_.reset(new RTPSender(
true, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
nullptr, false, nullptr, false, false));
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};
TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
class TestCallback : public StreamDataCountersCallback {
public:
@ -1663,94 +1623,6 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
rtp_sender_->RegisterRtpStatisticsCallback(nullptr);
}
TEST_P(RtpSenderAudioTest, SendAudio) {
const char payload_name[] = "PAYLOAD_NAME";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
}
TEST_P(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
kAudioLevelExtensionId));
const char payload_name[] = "PAYLOAD_NAME";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
auto sent_payload = transport_.last_sent_packet().payload();
EXPECT_THAT(sent_payload, ElementsAreArray(payload));
// Verify AudioLevel extension.
bool voice_activity;
uint8_t audio_level;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
&voice_activity, &audio_level));
EXPECT_EQ(kAudioLevel, audio_level);
EXPECT_FALSE(voice_activity);
}
// As RFC4733, named telephone events are carried as part of the audio stream
// and must use the same sequence number and timestamp base as the regular
// audio channel.
// This test checks the marker bit for the first packet and the consequent
// packets of the same telephone event. Since it is specifically for DTMF
// events, ignoring audio packets and sending kEmptyFrame instead of those.
TEST_P(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
const char* kDtmfPayloadName = "telephone-event";
const uint32_t kPayloadFrequency = 8000;
const uint8_t kPayloadType = 126;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(kDtmfPayloadName, kPayloadType,
kPayloadFrequency, 0, 0));
// For Telephone events, payload is not added to the registered payload list,
// it will register only the payload used for audio stream.
// Registering the payload again for audio stream with different payload name.
const char* kPayloadName = "payload_name";
ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType,
kPayloadFrequency, 1, 0));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
// DTMF event key=9, duration=500 and attenuationdB=10
rtp_sender_->SendTelephoneEvent(9, 500, 10);
// During start, it takes the starting timestamp as last sent timestamp.
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
RTPVideoHeader video_header;
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, kPayloadType, capture_time_ms, 0, nullptr, 0, nullptr,
&video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0, nullptr, 0, nullptr,
&video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// Marker Bit should be set to 1 for first packet.
EXPECT_TRUE(transport_.last_sent_packet().Marker());
ASSERT_TRUE(rtp_sender_->SendOutgoingData(
kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0, nullptr, 0, nullptr,
&video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
// Marker Bit should be set to 0 for rest of the packets.
EXPECT_FALSE(transport_.last_sent_packet().Marker());
}
TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
const char* kPayloadName = "GENERIC";
const uint8_t kPayloadType = 127;
@ -2295,7 +2167,5 @@ INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderVideoTest,
::testing::Bool());
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderAudioTest,
::testing::Bool());
} // namespace webrtc