Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(593)

Unified Diff: webrtc/video/video_send_stream_tests.cc

Issue 1542653002: Add audio streams to CallTest and a first A/V call test. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comment addressed Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/video/video_quality_test.cc ('k') | webrtc/video_engine_tests.isolate » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/video/video_send_stream_tests.cc
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 24ee2963d427c8c9cc7de376282a8821e08e090a..f0bac127dd4560f3e37f5d61922507de7fe6ed23 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -68,8 +68,8 @@ TEST_F(VideoSendStreamTest, CanStartStartedStream) {
CreateSenderCall(call_config);
test::NullTransport transport;
- CreateSendConfig(1, &transport);
- CreateStreams();
+ CreateSendConfig(1, 0, &transport);
+ CreateVideoStreams();
video_send_stream_->Start();
video_send_stream_->Start();
DestroyStreams();
@@ -80,8 +80,8 @@ TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
CreateSenderCall(call_config);
test::NullTransport transport;
- CreateSendConfig(1, &transport);
- CreateStreams();
+ CreateSendConfig(1, 0, &transport);
+ CreateVideoStreams();
video_send_stream_->Stop();
video_send_stream_->Stop();
DestroyStreams();
@@ -327,14 +327,14 @@ class FecObserver : public test::SendTest {
if (send_count_++ % 2 != 0) {
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics lossy_receive_stats(
- VideoSendStreamTest::kSendSsrcs[0], header.sequenceNumber,
+ VideoSendStreamTest::kVideoSendSsrcs[0], header.sequenceNumber,
send_count_ / 2, 127);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr,
transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(VideoSendStreamTest::kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(VideoSendStreamTest::kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -345,11 +345,12 @@ class FecObserver : public test::SendTest {
if (header.payloadType == VideoSendStreamTest::kRedPayloadType) {
encapsulated_payload_type = static_cast<int>(packet[header.headerLength]);
if (encapsulated_payload_type !=
- VideoSendStreamTest::kFakeSendPayloadType)
+ VideoSendStreamTest::kFakeVideoSendPayloadType)
EXPECT_EQ(VideoSendStreamTest::kUlpfecPayloadType,
encapsulated_payload_type);
} else {
- EXPECT_EQ(VideoSendStreamTest::kFakeSendPayloadType, header.payloadType);
+ EXPECT_EQ(VideoSendStreamTest::kFakeVideoSendPayloadType,
+ header.payloadType);
}
if (header_extensions_enabled_) {
@@ -459,7 +460,7 @@ void VideoSendStreamTest::TestNackRetransmission(
nullptr, transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -471,8 +472,8 @@ void VideoSendStreamTest::TestNackRetransmission(
uint16_t sequence_number = header.sequenceNumber;
if (header.ssrc == retransmit_ssrc_ &&
- retransmit_ssrc_ != kSendSsrcs[0]) {
- // Not kSendSsrcs[0], assume correct RTX packet. Extract sequence
+ retransmit_ssrc_ != kVideoSendSsrcs[0]) {
+ // Not kVideoSendSsrcs[0], assume correct RTX packet. Extract sequence
// number.
const uint8_t* rtx_header = packet + header.headerLength;
sequence_number = (rtx_header[0] << 8) + rtx_header[1];
@@ -496,7 +497,7 @@ void VideoSendStreamTest::TestNackRetransmission(
transport_adapter_->Enable();
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.rtx.payload_type = retransmit_payload_type_;
- if (retransmit_ssrc_ != kSendSsrcs[0])
+ if (retransmit_ssrc_ != kVideoSendSsrcs[0])
send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
}
@@ -516,7 +517,7 @@ void VideoSendStreamTest::TestNackRetransmission(
TEST_F(VideoSendStreamTest, RetransmitsNack) {
// Normal NACKs should use the send SSRC.
- TestNackRetransmission(kSendSsrcs[0], kFakeSendPayloadType);
+ TestNackRetransmission(kVideoSendSsrcs[0], kFakeVideoSendPayloadType);
}
TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
@@ -641,13 +642,13 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
if (packet_count_++ % 2 != 0) {
// Receive statistics reporting having lost 50% of the packets.
FakeReceiveStatistics lossy_receive_stats(
- kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
+ kVideoSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
&lossy_receive_stats, nullptr,
transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -864,13 +865,13 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
virtual void SendRtcpFeedback(int remb_value)
EXCLUSIVE_LOCKS_REQUIRED(crit_) {
- FakeReceiveStatistics receive_stats(
- kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
+ FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0],
+ last_sequence_number_, rtp_count_, 0);
RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr,
transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
if (remb_value > 0) {
rtcp_sender.SetREMBStatus(true);
rtcp_sender.SetREMBData(remb_value, std::vector<uint32_t>());
@@ -921,12 +922,12 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
kVideoMutedThresholdMs)
observation_complete_.Set();
// Receive statistics reporting having lost 50% of the packets.
- FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
+ FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0], 1, 1, 0);
RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), &receive_stats,
nullptr, transport_adapter_.get());
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
- rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
+ rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
RTCPSender::FeedbackState feedback_state;
@@ -942,7 +943,7 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
transport_adapter_->Enable();
}
- size_t GetNumStreams() const override { return 3; }
+ size_t GetNumVideoStreams() const override { return 3; }
virtual void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) {
@@ -1085,7 +1086,7 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
CreateSenderCall(Call::Config());
test::NullTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
Call::Config::BitrateConfig bitrate_config;
bitrate_config.start_bitrate_bps =
@@ -1095,7 +1096,7 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
StartBitrateObserver encoder;
video_send_config_.encoder_settings.encoder = &encoder;
- CreateStreams();
+ CreateVideoStreams();
EXPECT_EQ(video_encoder_config_.streams[0].max_bitrate_bps / 1000,
encoder.GetStartBitrateKbps());
@@ -1145,10 +1146,10 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
CreateSenderCall(Call::Config());
test::NullTransport transport;
- CreateSendConfig(1, &transport);
+ CreateSendConfig(1, 0, &transport);
FrameObserver observer;
video_send_config_.pre_encode_callback = &observer;
- CreateStreams();
+ CreateVideoStreams();
// Prepare five input frames. Send ordinary VideoFrame and texture frames
// alternatively.
@@ -1819,7 +1820,7 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
EXPECT_EQ(kNumStreams, encoder_config->streams.size());
}
- size_t GetNumStreams() const override { return kNumStreams; }
+ size_t GetNumVideoStreams() const override { return kNumStreams; }
void PerformTest() override {
EXPECT_TRUE(Wait())
@@ -1827,12 +1828,12 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
VideoSendStream::Stats stats = send_stream_->GetStats();
for (size_t i = 0; i < kNumStreams; ++i) {
- ASSERT_TRUE(stats.substreams.find(kSendSsrcs[i]) !=
+ ASSERT_TRUE(stats.substreams.find(kVideoSendSsrcs[i]) !=
stats.substreams.end())
- << "No stats for SSRC: " << kSendSsrcs[i]
+ << "No stats for SSRC: " << kVideoSendSsrcs[i]
<< ", stats should exist as soon as frames have been encoded.";
VideoSendStream::StreamStats ssrc_stats =
- stats.substreams[kSendSsrcs[i]];
+ stats.substreams[kVideoSendSsrcs[i]];
EXPECT_EQ(kEncodedResolution[i].width, ssrc_stats.width);
EXPECT_EQ(kEncodedResolution[i].height, ssrc_stats.height);
}
« no previous file with comments | « webrtc/video/video_quality_test.cc ('k') | webrtc/video_engine_tests.isolate » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698