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/*
* 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 <string>
#include <utility>
#include <vector>
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
namespace webrtc {
namespace test {
namespace {
enum : int { // The first valid value is 1.
kAudioLevelExtensionId = 1,
kTransportSequenceNumberExtensionId,
};
class AudioSendTest : public SendTest {
public:
AudioSendTest() : SendTest(CallTest::kDefaultTimeout) {}
size_t GetNumVideoStreams() const override { return 0; }
size_t GetNumAudioStreams() const override { return 1; }
size_t GetNumFlexfecStreams() const override { return 0; }
};
} // namespace
using AudioSendStreamCallTest = CallTest;
TEST_F(AudioSendStreamCallTest, SupportsCName) {
static std::string kCName = "PjqatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
class CNameObserver : public AudioSendTest {
public:
CNameObserver() = default;
private:
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
if (parser.sdes()->num_packets() > 0) {
EXPECT_EQ(1u, parser.sdes()->chunks().size());
EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);
observation_complete_.Set();
}
return SEND_PACKET;
}
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override {
send_config->rtp.c_name = kCName;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
}
} test;
RunBaseTest(&test);
}
TEST_F(AudioSendStreamCallTest, NoExtensionsByDefault) {
class NoExtensionsObserver : public AudioSendTest {
public:
NoExtensionsObserver() = default;
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length)); // rtp packet is valid.
EXPECT_EQ(packet[0] & 0b0001'0000, 0); // extension bit not set.
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override {
send_config->rtp.extensions.clear();
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
}
} test;
RunBaseTest(&test);
}
TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
class AudioLevelObserver : public AudioSendTest {
public:
AudioLevelObserver() : AudioSendTest() {
extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
uint8_t audio_level = 0;
bool voice = false;
EXPECT_TRUE(rtp_packet.GetExtension<AudioLevel>(&voice, &audio_level));
if (audio_level != 0) {
// Wait for at least one packet with a non-zero level.
observation_complete_.Set();
} else {
RTC_LOG(LS_WARNING) << "Got a packet with zero audioLevel - waiting"
" for another packet...";
}
return SEND_PACKET;
}
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelExtensionId));
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
}
private:
RtpHeaderExtensionMap extensions_;
} test;
RunBaseTest(&test);
}
class TransportWideSequenceNumberObserver : public AudioSendTest {
public:
explicit TransportWideSequenceNumberObserver(bool expect_sequence_number)
: AudioSendTest(), expect_sequence_number_(expect_sequence_number) {
extensions_.Register<TransportSequenceNumber>(
kTransportSequenceNumberExtensionId);
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet(&extensions_);
EXPECT_TRUE(rtp_packet.Parse(packet, length));
EXPECT_EQ(rtp_packet.HasExtension<TransportSequenceNumber>(),
expect_sequence_number_);
EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>());
EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>());
observation_complete_.Set();
return SEND_PACKET;
}
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
}
const bool expect_sequence_number_;
RtpHeaderExtensionMap extensions_;
};
TEST_F(AudioSendStreamCallTest, SendsTransportWideSequenceNumbersInFieldTrial) {
TransportWideSequenceNumberObserver test(/*expect_sequence_number=*/true);
RunBaseTest(&test);
}
TEST_F(AudioSendStreamCallTest, SendDtmf) {
static const uint8_t kDtmfPayloadType = 120;
static const int kDtmfPayloadFrequency = 8000;
static const int kDtmfEventFirst = 12;
static const int kDtmfEventLast = 31;
static const int kDtmfDuration = 50;
class DtmfObserver : public AudioSendTest {
public:
DtmfObserver() = default;
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
if (rtp_packet.PayloadType() == kDtmfPayloadType) {
EXPECT_EQ(rtp_packet.headers_size(), 12u);
EXPECT_EQ(rtp_packet.size(), 16u);
const int event = rtp_packet.payload()[0];
if (event != expected_dtmf_event_) {
++expected_dtmf_event_;
EXPECT_EQ(event, expected_dtmf_event_);
if (expected_dtmf_event_ == kDtmfEventLast) {
observation_complete_.Set();
}
}
}
return SEND_PACKET;
}
void OnAudioStreamsCreated(AudioSendStream* send_stream,
const std::vector<AudioReceiveStreamInterface*>&
receive_streams) override {
// Need to start stream here, else DTMF events are dropped.
send_stream->Start();
for (int event = kDtmfEventFirst; event <= kDtmfEventLast; ++event) {
send_stream->SendTelephoneEvent(kDtmfPayloadType, kDtmfPayloadFrequency,
event, kDtmfDuration);
}
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for DTMF stream.";
}
int expected_dtmf_event_ = kDtmfEventFirst;
} test;
RunBaseTest(&test);
}
} // namespace test
} // namespace webrtc
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