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Unified Diff: webrtc/call/bitrate_estimator_tests.cc

Issue 1537273003: Step 1 to prepare call_test.* for combined audio/video tests. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Cleanups Created 5 years ago
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Index: webrtc/call/bitrate_estimator_tests.cc
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index d8c0d5e34d33b22e0cc0d1cb6b6805514f1f648a..e371270df63c8510f5a764d788925457e4045fa3 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -119,17 +119,17 @@ class BitrateEstimatorTest : public test::CallTest {
receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
receive_transport_->SetReceiver(sender_call_->Receiver());
- send_config_ = VideoSendStream::Config(send_transport_.get());
- send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
+ video_send_config_ = VideoSendStream::Config(send_transport_.get());
+ video_send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
// Encoders will be set separately per stream.
- send_config_.encoder_settings.encoder = nullptr;
- send_config_.encoder_settings.payload_name = "FAKE";
- send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
- encoder_config_.streams = test::CreateVideoStreams(1);
+ video_send_config_.encoder_settings.encoder = nullptr;
+ video_send_config_.encoder_settings.payload_name = "FAKE";
+ video_send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
+ video_encoder_config_.streams = test::CreateVideoStreams(1);
receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
- receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
+ receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
receive_config_.rtp.remb = true;
receive_config_.rtp.extensions.push_back(
@@ -168,21 +168,21 @@ class BitrateEstimatorTest : public test::CallTest {
frame_generator_capturer_(),
fake_encoder_(Clock::GetRealTimeClock()),
fake_decoder_() {
- test_->send_config_.rtp.ssrcs[0]++;
- test_->send_config_.encoder_settings.encoder = &fake_encoder_;
+ test_->video_send_config_.rtp.ssrcs[0]++;
+ test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
send_stream_ = test_->sender_call_->CreateVideoSendStream(
- test_->send_config_, test_->encoder_config_);
- RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size());
+ test_->video_send_config_, test_->video_encoder_config_);
+ RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
- send_stream_->Input(), test_->encoder_config_.streams[0].width,
- test_->encoder_config_.streams[0].height, 30,
+ send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
+ test_->video_encoder_config_.streams[0].height, 30,
Clock::GetRealTimeClock()));
send_stream_->Start();
frame_generator_capturer_->Start();
if (receive_audio) {
AudioReceiveStream::Config receive_config;
- receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
+ receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
// Bogus non-default id to prevent hitting a RTC_DCHECK when creating
// the AudioReceiveStream. Every receive stream has to correspond to
// an underlying channel id.
@@ -196,13 +196,13 @@ class BitrateEstimatorTest : public test::CallTest {
VideoReceiveStream::Decoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type =
- test_->send_config_.encoder_settings.payload_type;
+ test_->video_send_config_.encoder_settings.payload_type;
decoder.payload_name =
- test_->send_config_.encoder_settings.payload_name;
+ test_->video_send_config_.encoder_settings.payload_name;
test_->receive_config_.decoders.clear();
test_->receive_config_.decoders.push_back(decoder);
test_->receive_config_.rtp.remote_ssrc =
- test_->send_config_.rtp.ssrcs[0];
+ test_->video_send_config_.rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_);
@@ -264,7 +264,7 @@ static const char* kSingleStreamLog =
"RemoteBitrateEstimatorSingleStream: Instantiating.";
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
@@ -273,7 +273,7 @@ TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
@@ -284,7 +284,7 @@ TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
@@ -300,7 +300,7 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
streams_.push_back(new Stream(this, true));
EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
@@ -309,14 +309,14 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
@@ -325,21 +325,21 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
}
TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
receiver_log_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
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