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Diffstat (limited to 'third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc')
-rw-r--r--third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc80
1 files changed, 38 insertions, 42 deletions
diff --git a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
index 3092e53c2d..4387aedf53 100644
--- a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
+++ b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
@@ -105,7 +105,7 @@ class RtpSenderReceiverTest
: network_thread_(rtc::Thread::Current()),
worker_thread_(rtc::Thread::Current()),
video_bitrate_allocator_factory_(
- webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
+ CreateBuiltinVideoBitrateAllocatorFactory()),
// Create fake media engine/etc. so we can create channels to use to
// test RtpSenders/RtpReceivers.
media_engine_(std::make_unique<cricket::FakeMediaEngine>()),
@@ -119,16 +119,16 @@ class RtpSenderReceiverTest
// Fake media channels are owned by the media engine.
voice_media_send_channel_ = media_engine_->voice().CreateSendChannel(
&fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
- webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
+ CryptoOptions(), AudioCodecPairId::Create());
video_media_send_channel_ = media_engine_->video().CreateSendChannel(
&fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
- webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get());
+ CryptoOptions(), video_bitrate_allocator_factory_.get());
voice_media_receive_channel_ = media_engine_->voice().CreateReceiveChannel(
&fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
- webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
+ CryptoOptions(), AudioCodecPairId::Create());
video_media_receive_channel_ = media_engine_->video().CreateReceiveChannel(
&fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
- webrtc::CryptoOptions());
+ CryptoOptions());
// Create streams for predefined SSRCs. Streams need to exist in order
// for the senders and receievers to apply parameters to them.
@@ -162,8 +162,8 @@ class RtpSenderReceiverTest
audio_track_ = nullptr;
}
- std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
- auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
+ std::unique_ptr<RtpTransportInternal> CreateDtlsSrtpTransport() {
+ auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>(
/*rtcp_mux_required=*/true, field_trials_);
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
/*rtcp_dtls_transport=*/nullptr);
@@ -515,12 +515,12 @@ class RtpSenderReceiverTest
test::RunLoop run_loop_;
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
- webrtc::RtcEventLogNull event_log_;
+ RtcEventLogNull 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_;
- std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
+ std::unique_ptr<RtpTransportInternal> rtp_transport_;
+ std::unique_ptr<VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
std::unique_ptr<cricket::FakeMediaEngine> media_engine_;
rtc::UniqueRandomIdGenerator ssrc_generator_;
@@ -540,7 +540,7 @@ class RtpSenderReceiverTest
rtc::scoped_refptr<MediaStreamInterface> local_stream_;
rtc::scoped_refptr<VideoTrackInterface> video_track_;
rtc::scoped_refptr<AudioTrackInterface> audio_track_;
- webrtc::test::ScopedKeyValueConfig field_trials_;
+ test::ScopedKeyValueConfig field_trials_;
};
// Test that `voice_channel_` is updated when an audio track is associated
@@ -651,15 +651,13 @@ TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) {
TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
CreateVideoRtpReceiver();
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state());
- EXPECT_EQ(webrtc::MediaSourceInterface::kLive,
- video_track_->GetSource()->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state());
+ EXPECT_EQ(MediaSourceInterface::kLive, video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
- EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
- video_track_->GetSource()->state());
+ EXPECT_EQ(MediaStreamTrackInterface::kEnded, video_track_->state());
+ EXPECT_EQ(MediaSourceInterface::kEnded, video_track_->GetSource()->state());
DestroyVideoRtpReceiver();
}
@@ -888,9 +886,9 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersAsync) {
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -918,13 +916,13 @@ TEST_F(RtpSenderReceiverTest,
audio_rtp_sender_ =
AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
RtpParameters params = audio_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -932,7 +930,7 @@ TEST_F(RtpSenderReceiverTest,
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -1016,13 +1014,13 @@ TEST_F(RtpSenderReceiverTest,
RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
audio_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
@@ -1081,7 +1079,7 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
CreateAudioRtpSender();
EXPECT_EQ(-1, voice_media_send_channel()->max_bps());
- webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+ RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
params.encodings[0].max_bitrate_bps = 1000;
@@ -1106,10 +1104,9 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
CreateAudioRtpSender();
- webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+ RtpParameters params = audio_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
- EXPECT_EQ(webrtc::kDefaultBitratePriority,
- params.encodings[0].bitrate_priority);
+ EXPECT_EQ(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());
@@ -1140,9 +1137,9 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersAsync) {
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -1170,19 +1167,19 @@ TEST_F(RtpSenderReceiverTest,
video_rtp_sender_ =
VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
RtpParameters params = video_rtp_sender_->GetParameters();
ASSERT_EQ(1u, params.encodings.size());
params.encodings[0].max_bitrate_bps = 90000;
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
params = video_rtp_sender_->GetParameters();
EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
@@ -1350,13 +1347,13 @@ TEST_F(RtpSenderReceiverTest,
RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1u, params.encodings.size());
- absl::optional<webrtc::RTCError> result;
+ absl::optional<RTCError> result;
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_TRUE(result->ok());
video_rtp_sender_->SetParametersAsync(
- params, [&result](webrtc::RTCError error) { result = error; });
+ params, [&result](RTCError error) { result = error; });
run_loop_.Flush();
EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
@@ -1453,7 +1450,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidNumTemporalLayers) {
CreateVideoRtpSender();
RtpParameters params = video_rtp_sender_->GetParameters();
- params.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1;
+ params.encodings[0].num_temporal_layers = kMaxTemporalStreams + 1;
RTCError result = video_rtp_sender_->SetParameters(params);
EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type());
@@ -1536,7 +1533,7 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
CreateVideoRtpSender();
EXPECT_EQ(-1, video_media_send_channel()->max_bps());
- webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+ RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
@@ -1589,10 +1586,9 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) {
TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
CreateVideoRtpSender();
- webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+ RtpParameters params = video_rtp_sender_->GetParameters();
EXPECT_EQ(1U, params.encodings.size());
- EXPECT_EQ(webrtc::kDefaultBitratePriority,
- params.encodings[0].bitrate_priority);
+ EXPECT_EQ(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());
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-22 06:09:57 +0000