blob: 8151ad6bd58aaa004f01716c7c94fefadf73ce06 [file] [log] [blame]
/*
* Copyright 2012 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 <string>
#include <utility>
#include "api/rtpparameters.h"
#include "media/base/fakemediaengine.h"
#include "media/base/rtpdataengine.h"
#include "media/base/testutils.h"
#include "media/engine/fakewebrtccall.h"
#include "p2p/base/fakedtlstransport.h"
#include "pc/audiotrack.h"
#include "pc/channelmanager.h"
#include "pc/localaudiosource.h"
#include "pc/mediastream.h"
#include "pc/remoteaudiosource.h"
#include "pc/rtpreceiver.h"
#include "pc/rtpsender.h"
#include "pc/streamcollection.h"
#include "pc/test/fakevideotracksource.h"
#include "pc/videotrack.h"
#include "pc/videotracksource.h"
#include "rtc_base/gunit.h"
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::_;
using ::testing::Exactly;
using ::testing::InvokeWithoutArgs;
using ::testing::Return;
namespace {
static const char kStreamId1[] = "local_stream_1";
static const char kVideoTrackId[] = "video_1";
static const char kAudioTrackId[] = "audio_1";
static const uint32_t kVideoSsrc = 98;
static const uint32_t kVideoSsrc2 = 100;
static const uint32_t kAudioSsrc = 99;
static const uint32_t kAudioSsrc2 = 101;
static const int kDefaultTimeout = 10000; // 10 seconds.
} // namespace
namespace webrtc {
class RtpSenderReceiverTest : public testing::Test,
public sigslot::has_slots<> {
public:
RtpSenderReceiverTest()
: network_thread_(rtc::Thread::Current()),
worker_thread_(rtc::Thread::Current()),
// Create fake media engine/etc. so we can create channels to use to
// test RtpSenders/RtpReceivers.
media_engine_(new cricket::FakeMediaEngine()),
channel_manager_(rtc::WrapUnique(media_engine_),
rtc::MakeUnique<cricket::RtpDataEngine>(),
worker_thread_,
network_thread_),
fake_call_(),
local_stream_(MediaStream::Create(kStreamId1)) {
// Create channels to be used by the RtpSenders and RtpReceivers.
channel_manager_.Init();
bool srtp_required = true;
rtp_dtls_transport_ = rtc::MakeUnique<cricket::FakeDtlsTransport>(
"fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP);
rtp_transport_ = CreateDtlsSrtpTransport();
voice_channel_ = channel_manager_.CreateVoiceChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
rtc::Thread::Current(), cricket::CN_AUDIO, srtp_required,
rtc::CryptoOptions(), cricket::AudioOptions());
video_channel_ = channel_manager_.CreateVideoChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
rtc::Thread::Current(), cricket::CN_VIDEO, srtp_required,
rtc::CryptoOptions(), cricket::VideoOptions());
voice_channel_->Enable(true);
video_channel_->Enable(true);
voice_media_channel_ = media_engine_->GetVoiceChannel(0);
video_media_channel_ = media_engine_->GetVideoChannel(0);
RTC_CHECK(voice_channel_);
RTC_CHECK(video_channel_);
RTC_CHECK(voice_media_channel_);
RTC_CHECK(video_media_channel_);
// Create streams for predefined SSRCs. Streams need to exist in order
// for the senders and receievers to apply parameters to them.
// Normally these would be created by SetLocalDescription and
// SetRemoteDescription.
voice_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc));
voice_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc));
voice_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc2));
voice_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc2));
video_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc));
video_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc));
video_media_channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc2));
video_media_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kVideoSsrc2));
}
std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
auto dtls_srtp_transport =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(/*rtcp_mux_required=*/true);
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
/*rtcp_dtls_transport=*/nullptr);
return dtls_srtp_transport;
}
// Needed to use DTMF sender.
void AddDtmfCodec() {
cricket::AudioSendParameters params;
const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000,
0, 1);
params.codecs.push_back(kTelephoneEventCodec);
voice_media_channel_->SetSendParameters(params);
}
void AddVideoTrack() { AddVideoTrack(false); }
void AddVideoTrack(bool is_screencast) {
rtc::scoped_refptr<VideoTrackSourceInterface> source(
FakeVideoTrackSource::Create(is_screencast));
video_track_ =
VideoTrack::Create(kVideoTrackId, source, rtc::Thread::Current());
EXPECT_TRUE(local_stream_->AddTrack(video_track_));
}
void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
void CreateAudioRtpSender(
const rtc::scoped_refptr<LocalAudioSource>& source) {
audio_track_ = AudioTrack::Create(kAudioTrackId, source);
EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
audio_rtp_sender_ =
new AudioRtpSender(worker_thread_, local_stream_->GetAudioTracks()[0],
{local_stream_->id()}, nullptr);
audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
audio_rtp_sender_->GetOnDestroyedSignal()->connect(
this, &RtpSenderReceiverTest::OnAudioSenderDestroyed);
VerifyVoiceChannelInput();
}
void CreateAudioRtpSenderWithNoTrack() {
audio_rtp_sender_ = new AudioRtpSender(worker_thread_, nullptr);
audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
}
void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
void CreateVideoRtpSender(uint32_t ssrc) {
CreateVideoRtpSender(false, ssrc);
}
void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
void CreateVideoRtpSender(bool is_screencast, uint32_t ssrc = kVideoSsrc) {
AddVideoTrack(is_screencast);
video_rtp_sender_ =
new VideoRtpSender(worker_thread_, local_stream_->GetVideoTracks()[0],
{local_stream_->id()});
video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
video_rtp_sender_->SetSsrc(ssrc);
VerifyVideoChannelInput(ssrc);
}
void CreateVideoRtpSenderWithNoTrack() {
video_rtp_sender_ = new VideoRtpSender(worker_thread_);
video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
}
void DestroyAudioRtpSender() {
audio_rtp_sender_ = nullptr;
VerifyVoiceChannelNoInput();
}
void DestroyVideoRtpSender() {
video_rtp_sender_ = nullptr;
VerifyVideoChannelNoInput();
}
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
audio_rtp_receiver_ = new AudioRtpReceiver(
rtc::Thread::Current(), kAudioTrackId, std::move(streams));
audio_rtp_receiver_->SetVoiceMediaChannel(voice_media_channel_);
audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
audio_track_ = audio_rtp_receiver_->audio_track();
VerifyVoiceChannelOutput();
}
void CreateVideoRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
video_rtp_receiver_ = new VideoRtpReceiver(
rtc::Thread::Current(), kVideoTrackId, std::move(streams));
video_rtp_receiver_->SetVideoMediaChannel(video_media_channel_);
video_rtp_receiver_->SetupMediaChannel(kVideoSsrc);
video_track_ = video_rtp_receiver_->video_track();
VerifyVideoChannelOutput();
}
void DestroyAudioRtpReceiver() {
audio_rtp_receiver_ = nullptr;
VerifyVoiceChannelNoOutput();
}
void DestroyVideoRtpReceiver() {
video_rtp_receiver_ = nullptr;
VerifyVideoChannelNoOutput();
}
void VerifyVoiceChannelInput() { VerifyVoiceChannelInput(kAudioSsrc); }
void VerifyVoiceChannelInput(uint32_t ssrc) {
// Verify that the media channel has an audio source, and the stream isn't
// muted.
EXPECT_TRUE(voice_media_channel_->HasSource(ssrc));
EXPECT_FALSE(voice_media_channel_->IsStreamMuted(ssrc));
}
void VerifyVideoChannelInput() { VerifyVideoChannelInput(kVideoSsrc); }
void VerifyVideoChannelInput(uint32_t ssrc) {
// Verify that the media channel has a video source,
EXPECT_TRUE(video_media_channel_->HasSource(ssrc));
}
void VerifyVoiceChannelNoInput() { VerifyVoiceChannelNoInput(kAudioSsrc); }
void VerifyVoiceChannelNoInput(uint32_t ssrc) {
// Verify that the media channel's source is reset.
EXPECT_FALSE(voice_media_channel_->HasSource(ssrc));
}
void VerifyVideoChannelNoInput() { VerifyVideoChannelNoInput(kVideoSsrc); }
void VerifyVideoChannelNoInput(uint32_t ssrc) {
// Verify that the media channel's source is reset.
EXPECT_FALSE(video_media_channel_->HasSource(ssrc));
}
void VerifyVoiceChannelOutput() {
// Verify that the volume is initialized to 1.
double volume;
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
}
void VerifyVideoChannelOutput() {
// Verify that the media channel has a sink.
EXPECT_TRUE(video_media_channel_->HasSink(kVideoSsrc));
}
void VerifyVoiceChannelNoOutput() {
// Verify that the volume is reset to 0.
double volume;
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
}
void VerifyVideoChannelNoOutput() {
// Verify that the media channel's sink is reset.
EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc));
}
protected:
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
webrtc::RtcEventLogNullImpl event_log_;
// The |rtp_dtls_transport_| and |rtp_transport_| should be destroyed after
// the |channel_manager|.
std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;
std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
// |media_engine_| is actually owned by |channel_manager_|.
cricket::FakeMediaEngine* media_engine_;
cricket::ChannelManager channel_manager_;
cricket::FakeCall fake_call_;
cricket::VoiceChannel* voice_channel_;
cricket::VideoChannel* video_channel_;
cricket::FakeVoiceMediaChannel* voice_media_channel_;
cricket::FakeVideoMediaChannel* video_media_channel_;
rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_;
rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_;
rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_;
rtc::scoped_refptr<MediaStreamInterface> local_stream_;
rtc::scoped_refptr<VideoTrackInterface> video_track_;
rtc::scoped_refptr<AudioTrackInterface> audio_track_;
bool audio_sender_destroyed_signal_fired_ = false;
};
// Test that |voice_channel_| is updated when an audio track is associated
// and disassociated with an AudioRtpSender.
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) {
CreateAudioRtpSender();
DestroyAudioRtpSender();
}
// Test that |video_channel_| is updated when a video track is associated and
// disassociated with a VideoRtpSender.
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) {
CreateVideoRtpSender();
DestroyVideoRtpSender();
}
// Test that |voice_channel_| is updated when a remote audio track is
// associated and disassociated with an AudioRtpReceiver.
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) {
CreateAudioRtpReceiver();
DestroyAudioRtpReceiver();
}
// Test that |video_channel_| is updated when a remote video track is
// associated and disassociated with a VideoRtpReceiver.
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) {
CreateVideoRtpReceiver();
DestroyVideoRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiverWithStreams) {
CreateAudioRtpReceiver({local_stream_});
DestroyAudioRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiverWithStreams) {
CreateVideoRtpReceiver({local_stream_});
DestroyVideoRtpReceiver();
}
// Test that the AudioRtpSender applies options from the local audio source.
TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) {
cricket::AudioOptions options;
options.echo_cancellation = true;
auto source = LocalAudioSource::Create(&options);
CreateAudioRtpSender(source.get());
EXPECT_EQ(true, voice_media_channel_->options().echo_cancellation);
DestroyAudioRtpSender();
}
// Test that the stream is muted when the track is disabled, and unmuted when
// the track is enabled.
TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) {
CreateAudioRtpSender();
audio_track_->set_enabled(false);
EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc));
audio_track_->set_enabled(true);
EXPECT_FALSE(voice_media_channel_->IsStreamMuted(kAudioSsrc));
DestroyAudioRtpSender();
}
// Test that the volume is set to 0 when the track is disabled, and back to
// 1 when the track is enabled.
TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) {
CreateAudioRtpReceiver();
double volume;
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
audio_track_->set_enabled(false);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
audio_track_->set_enabled(true);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(1, volume);
DestroyAudioRtpReceiver();
}
// Currently no action is taken when a remote video track is disabled or
// enabled, so there's nothing to test here, other than what is normally
// verified in DestroyVideoRtpSender.
TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) {
CreateVideoRtpSender();
video_track_->set_enabled(false);
video_track_->set_enabled(true);
DestroyVideoRtpSender();
}
// Test that the state of the video track created by the VideoRtpReceiver is
// updated when the receiver is destroyed.
TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
CreateVideoRtpReceiver();
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state());
EXPECT_EQ(webrtc::MediaSourceInterface::kLive,
video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
video_track_->GetSource()->state());
}
// Currently no action is taken when a remote video track is disabled or
// enabled, so there's nothing to test here, other than what is normally
// verified in DestroyVideoRtpReceiver.
TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) {
CreateVideoRtpReceiver();
video_track_->set_enabled(false);
video_track_->set_enabled(true);
DestroyVideoRtpReceiver();
}
// Test that the AudioRtpReceiver applies volume changes from the track source
// to the media channel.
TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) {
CreateAudioRtpReceiver();
double volume;
audio_track_->GetSource()->SetVolume(0.5);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.5, volume);
// Disable the audio track, this should prevent setting the volume.
audio_track_->set_enabled(false);
audio_track_->GetSource()->SetVolume(0.8);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0, volume);
// When the track is enabled, the previously set volume should take effect.
audio_track_->set_enabled(true);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.8, volume);
// Try changing volume one more time.
audio_track_->GetSource()->SetVolume(0.9);
EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
EXPECT_EQ(0.9, volume);
DestroyAudioRtpReceiver();
}
// Test that the media channel isn't enabled for sending if the audio sender
// doesn't have both a track and SSRC.
TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
// Track but no SSRC.
EXPECT_TRUE(audio_rtp_sender_->SetTrack(track));
VerifyVoiceChannelNoInput();
// SSRC but no track.
EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr));
audio_rtp_sender_->SetSsrc(kAudioSsrc);
VerifyVoiceChannelNoInput();
}
// Test that the media channel isn't enabled for sending if the video sender
// doesn't have both a track and SSRC.
TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
CreateVideoRtpSenderWithNoTrack();
// Track but no SSRC.
EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_));
VerifyVideoChannelNoInput();
// SSRC but no track.
EXPECT_TRUE(video_rtp_sender_->SetTrack(nullptr));
video_rtp_sender_->SetSsrc(kVideoSsrc);
VerifyVideoChannelNoInput();
}
// Test that the media channel is enabled for sending when the audio sender
// has a track and SSRC, when the SSRC is set first.
TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
audio_rtp_sender_->SetTrack(track);
VerifyVoiceChannelInput();
DestroyAudioRtpSender();
}
// Test that the media channel is enabled for sending when the audio sender
// has a track and SSRC, when the SSRC is set last.
TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
CreateAudioRtpSenderWithNoTrack();
rtc::scoped_refptr<AudioTrackInterface> track =
AudioTrack::Create(kAudioTrackId, nullptr);
audio_rtp_sender_->SetTrack(track);
audio_rtp_sender_->SetSsrc(kAudioSsrc);
VerifyVoiceChannelInput();
DestroyAudioRtpSender();
}
// Test that the media channel is enabled for sending when the video sender
// has a track and SSRC, when the SSRC is set first.
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
AddVideoTrack();
CreateVideoRtpSenderWithNoTrack();
video_rtp_sender_->SetSsrc(kVideoSsrc);
video_rtp_sender_->SetTrack(video_track_);
VerifyVideoChannelInput();
DestroyVideoRtpSender();
}
// Test that the media channel is enabled for sending when the video sender
// has a track and SSRC, when the SSRC is set last.
TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
AddVideoTrack();
CreateVideoRtpSenderWithNoTrack();
video_rtp_sender_->SetTrack(video_track_);
video_rtp_sender_->SetSsrc(kVideoSsrc);
VerifyVideoChannelInput();
DestroyVideoRtpSender();
}
// Test that the media channel stops sending when the audio sender's SSRC is set
// to 0.
TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
CreateAudioRtpSender();
audio_rtp_sender_->SetSsrc(0);
VerifyVoiceChannelNoInput();
}
// Test that the media channel stops sending when the video sender's SSRC is set
// to 0.
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
CreateAudioRtpSender();
audio_rtp_sender_->SetSsrc(0);
VerifyVideoChannelNoInput();
}
// Test that the media channel stops sending when the audio sender's track is
// set to null.
TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
CreateAudioRtpSender();
EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr));
VerifyVoiceChannelNoInput();
}
// Test that the media channel stops sending when the video sender's track is
// set to null.
TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
CreateVideoRtpSender();
video_rtp_sender_->SetSsrc(0);
VerifyVideoChannelNoInput();
}
// Test that when the audio sender's SSRC is changed, the media channel stops
// sending with the old SSRC and starts sending with the new one.
TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
CreateAudioRtpSender();
audio_rtp_sender_->SetSsrc(kAudioSsrc2);
VerifyVoiceChannelNoInput(kAudioSsrc);
VerifyVoiceChannelInput(kAudioSsrc2);
audio_rtp_sender_ = nullptr;
VerifyVoiceChannelNoInput(kAudioSsrc2);
}
// Test that when the audio sender's SSRC is changed, the media channel stops
// sending with the old SSRC and starts sending with the new one.
TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
CreateVideoRtpSender();
video_rtp_sender_->SetSsrc(kVideoSsrc2);
VerifyVideoChannelNoInput(kVideoSsrc);
VerifyVideoChannelInput(kVideoSsrc2);
video_rtp_sender_ = nullptr;
VerifyVideoChannelNoInput(kVideoSsrc2);
}
TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderMustCallGetParametersBeforeSetParameters) {
CreateAudioRtpSender();
RtpParameters params;
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderSetParametersInvalidatesTransactionId) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderDetectTransactionIdModification) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
params.transaction_id = "";
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderCheckTransactionIdRefresh) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_NE(params.transaction_id.size(), 0);
auto saved_transaction_id = params.transaction_id;
params = audio_rtp_sender_->GetParameters();
EXPECT_NE(saved_transaction_id, params.transaction_id);
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderSetParametersOldValueFail) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
RtpParameters second_params = audio_rtp_sender_->GetParameters();
RTCError result = audio_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderCantSetUnimplementedRtpParameters) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
// Unimplemented RtpParameters: mid, header_extensions,
// degredation_preference.
params.mid = "dummy_mid";
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest,
AudioSenderCantSetUnimplementedRtpEncodingParameters) {
CreateAudioRtpSender();
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
// Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
// max_framerate, scale_resolution_down_by, scale_framerate_down_by, rid,
// dependency_rids.
params.encodings[0].codec_payload_type = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].fec = RtpFecParameters();
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].rtx = RtpRtxParameters();
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].dtx = DtxStatus::ENABLED;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].ptime = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].max_framerate = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].scale_resolution_down_by = 2.0;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].rid = "dummy_rid";
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
params = audio_rtp_sender_->GetParameters();
params.encodings[0].dependency_rids.push_back("dummy_rid");
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
audio_rtp_sender_->SetParameters(params).type());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
CreateAudioRtpSender();
EXPECT_EQ(-1, voice_media_channel_->max_bps());
webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
params.encodings[0].max_bitrate_bps = 1000;
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
// Read back the parameters and verify they have been changed.
params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the audio channel received the new parameters.
params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the global bitrate limit has not been changed.
EXPECT_EQ(-1, voice_media_channel_->max_bps());
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
CreateAudioRtpSender();
webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
params.encodings[0].bitrate_priority);
double new_bitrate_priority = 2.0;
params.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
DestroyAudioRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderMustCallGetParametersBeforeSetParameters) {
CreateVideoRtpSender();
RtpParameters params;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderSetParametersInvalidatesTransactionId) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderDetectTransactionIdModification) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
params.transaction_id = "";
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCheckTransactionIdRefresh) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_NE(params.transaction_id.size(), 0);
auto saved_transaction_id = params.transaction_id;
params = video_rtp_sender_->GetParameters();
EXPECT_NE(saved_transaction_id, params.transaction_id);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderSetParametersOldValueFail) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
RtpParameters second_params = video_rtp_sender_->GetParameters();
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, VideoSenderCantSetUnimplementedRtpParameters) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
// Unimplemented RtpParameters: mid, header_extensions,
// degredation_preference.
params.mid = "dummy_mid";
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest,
VideoSenderCantSetUnimplementedEncodingParameters) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
// Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
// max_framerate, scale_resolution_down_by, scale_framerate_down_by, rid,
// dependency_rids.
params.encodings[0].codec_payload_type = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].fec = RtpFecParameters();
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].rtx = RtpRtxParameters();
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].dtx = DtxStatus::ENABLED;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].ptime = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].max_framerate = 1;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].scale_resolution_down_by = 2.0;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].rid = "dummy_rid";
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
params.encodings[0].dependency_rids.push_back("dummy_rid");
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
DestroyVideoRtpSender();
}
// A video sender can have multiple simulcast layers, in which case it will
// contain multiple RtpEncodingParameters. This tests that if this is the case
// (simulcast), then we can't set the bitrate_priority, or max_bitrate_bps
// for any encodings besides at index 0, because these are both implemented
// "per-sender."
TEST_F(RtpSenderReceiverTest, VideoSenderCantSetPerSenderEncodingParameters) {
// Add a simulcast specific send stream that contains 2 encoding parameters.
std::vector<uint32_t> ssrcs({1, 2});
cricket::StreamParams stream_params =
cricket::CreateSimStreamParams("cname", ssrcs);
video_media_channel_->AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
CreateVideoRtpSender(primary_ssrc);
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(ssrcs.size(), params.encodings.size());
params.encodings[1].bitrate_priority = 2.0;
EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
video_rtp_sender_->SetParameters(params).type());
params = video_rtp_sender_->GetParameters();
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
CreateVideoRtpSender();
EXPECT_EQ(-1, video_media_channel_->max_bps());
webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
params.encodings[0].min_bitrate_bps = 100;
params.encodings[0].max_bitrate_bps = 1000;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
// Read back the parameters and verify they have been changed.
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the video channel received the new parameters.
params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
// Verify that the global bitrate limit has not been changed.
EXPECT_EQ(-1, video_media_channel_->max_bps());
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) {
// Add a simulcast specific send stream that contains 2 encoding parameters.
std::vector<uint32_t> ssrcs({1, 2});
cricket::StreamParams stream_params =
cricket::CreateSimStreamParams("cname", ssrcs);
video_media_channel_->AddSendStream(stream_params);
uint32_t primary_ssrc = stream_params.first_ssrc();
CreateVideoRtpSender(primary_ssrc);
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(ssrcs.size(), params.encodings.size());
params.encodings[0].min_bitrate_bps = 100;
params.encodings[0].max_bitrate_bps = 1000;
params.encodings[1].min_bitrate_bps = 200;
params.encodings[1].max_bitrate_bps = 2000;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
// Verify that the video channel received the new parameters.
params = video_media_channel_->GetRtpSendParameters(primary_ssrc);
EXPECT_EQ(ssrcs.size(), params.encodings.size());
EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
EXPECT_EQ(200, params.encodings[1].min_bitrate_bps);
EXPECT_EQ(2000, params.encodings[1].max_bitrate_bps);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
CreateVideoRtpSender();
webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(webrtc::kDefaultBitratePriority,
params.encodings[0].bitrate_priority);
double new_bitrate_priority = 2.0;
params.encodings[0].bitrate_priority = new_bitrate_priority;
EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
EXPECT_EQ(1, params.encodings.size());
EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
CreateAudioRtpReceiver();
RtpParameters params = audio_rtp_receiver_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
DestroyAudioRtpReceiver();
}
TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
CreateVideoRtpReceiver();
RtpParameters params = video_rtp_receiver_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
DestroyVideoRtpReceiver();
}
// Test that makes sure that a video track content hint translates to the proper
// value for sources that are not screencast.
TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) {
CreateVideoRtpSender();
video_track_->set_enabled(true);
// |video_track_| is not screencast by default.
EXPECT_EQ(false, video_media_channel_->options().is_screencast);
// No content hint should be set by default.
EXPECT_EQ(VideoTrackInterface::ContentHint::kNone,
video_track_->content_hint());
// Setting detailed should turn a non-screencast source into screencast mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
EXPECT_EQ(true, video_media_channel_->options().is_screencast);
// Removing the content hint should turn the track back into non-screencast
// mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
EXPECT_EQ(false, video_media_channel_->options().is_screencast);
// Setting fluid should remain in non-screencast mode (its default).
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid);
EXPECT_EQ(false, video_media_channel_->options().is_screencast);
DestroyVideoRtpSender();
}
// Test that makes sure that a video track content hint translates to the proper
// value for screencast sources.
TEST_F(RtpSenderReceiverTest,
PropagatesVideoTrackContentHintForScreencastSource) {
CreateVideoRtpSender(true);
video_track_->set_enabled(true);
// |video_track_| with a screencast source should be screencast by default.
EXPECT_EQ(true, video_media_channel_->options().is_screencast);
// No content hint should be set by default.
EXPECT_EQ(VideoTrackInterface::ContentHint::kNone,
video_track_->content_hint());
// Setting fluid should turn a screencast source into non-screencast mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid);
EXPECT_EQ(false, video_media_channel_->options().is_screencast);
// Removing the content hint should turn the track back into screencast mode.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
EXPECT_EQ(true, video_media_channel_->options().is_screencast);
// Setting detailed should still remain in screencast mode (its default).
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
EXPECT_EQ(true, video_media_channel_->options().is_screencast);
DestroyVideoRtpSender();
}
// Test that makes sure any content hints that are set on a track before
// VideoRtpSender is ready to send are still applied when it gets ready to send.
TEST_F(RtpSenderReceiverTest,
PropagatesVideoTrackContentHintSetBeforeEnabling) {
AddVideoTrack();
// Setting detailed overrides the default non-screencast mode. This should be
// applied even if the track is set on construction.
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
video_rtp_sender_ =
new VideoRtpSender(worker_thread_, local_stream_->GetVideoTracks()[0],
{local_stream_->id()});
video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
video_track_->set_enabled(true);
// Sender is not ready to send (no SSRC) so no option should have been set.
EXPECT_EQ(absl::nullopt, video_media_channel_->options().is_screencast);
// Verify that the content hint is accounted for when video_rtp_sender_ does
// get enabled.
video_rtp_sender_->SetSsrc(kVideoSsrc);
EXPECT_EQ(true, video_media_channel_->options().is_screencast);
// And removing the hint should go back to false (to verify that false was
// default correctly).
video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone);
EXPECT_EQ(false, video_media_channel_->options().is_screencast);
DestroyVideoRtpSender();
}
TEST_F(RtpSenderReceiverTest, AudioSenderHasDtmfSender) {
CreateAudioRtpSender();
EXPECT_NE(nullptr, audio_rtp_sender_->GetDtmfSender());
}
TEST_F(RtpSenderReceiverTest, VideoSenderDoesNotHaveDtmfSender) {
CreateVideoRtpSender();
EXPECT_EQ(nullptr, video_rtp_sender_->GetDtmfSender());
}
// Test that the DTMF sender is really using |voice_channel_|, and thus returns
// true/false from CanSendDtmf based on what |voice_channel_| returns.
TEST_F(RtpSenderReceiverTest, CanInsertDtmf) {
AddDtmfCodec();
CreateAudioRtpSender();
auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
ASSERT_NE(nullptr, dtmf_sender);
EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
}
TEST_F(RtpSenderReceiverTest, CanNotInsertDtmf) {
CreateAudioRtpSender();
auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
ASSERT_NE(nullptr, dtmf_sender);
// DTMF codec has not been added, as it was in the above test.
EXPECT_FALSE(dtmf_sender->CanInsertDtmf());
}
TEST_F(RtpSenderReceiverTest, InsertDtmf) {
AddDtmfCodec();
CreateAudioRtpSender();
auto dtmf_sender = audio_rtp_sender_->GetDtmfSender();
ASSERT_NE(nullptr, dtmf_sender);
EXPECT_EQ(0U, voice_media_channel_->dtmf_info_queue().size());
// Insert DTMF
const int expected_duration = 90;
dtmf_sender->InsertDtmf("012", expected_duration, 100);
// Verify
ASSERT_EQ_WAIT(3U, voice_media_channel_->dtmf_info_queue().size(),
kDefaultTimeout);
const uint32_t send_ssrc =
voice_media_channel_->send_streams()[0].first_ssrc();
EXPECT_TRUE(CompareDtmfInfo(voice_media_channel_->dtmf_info_queue()[0],
send_ssrc, 0, expected_duration));
EXPECT_TRUE(CompareDtmfInfo(voice_media_channel_->dtmf_info_queue()[1],
send_ssrc, 1, expected_duration));
EXPECT_TRUE(CompareDtmfInfo(voice_media_channel_->dtmf_info_queue()[2],
send_ssrc, 2, expected_duration));
}
// Make sure the signal from "GetOnDestroyedSignal()" fires when the sender is
// destroyed, which is needed for the DTMF sender.
TEST_F(RtpSenderReceiverTest, TestOnDestroyedSignal) {
CreateAudioRtpSender();
EXPECT_FALSE(audio_sender_destroyed_signal_fired_);
audio_rtp_sender_ = nullptr;
EXPECT_TRUE(audio_sender_destroyed_signal_fired_);
}
} // namespace webrtc