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Unified Diff: webrtc/video/video_send_stream_tests.cc

Issue 2366333003: Fix race / crash in OnNetworkRouteChanged(). (Closed)
Patch Set: Created 4 years, 3 months ago
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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 ba15b17c85b229604bd7ba082e836b00ac54fc4f..04b954ac12ef0d47e15c1972ccf2ce09e700b7e2 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1136,19 +1136,43 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
}
TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
+ static const int kStartBitrateBps = 300000;
+ static const int kNewMaxBitrateBps = 1234567;
+ static const uint8_t kExtensionId = 13;
class ChangingNetworkRouteTest : public test::EndToEndTest {
public:
- const int kStartBitrateBps = 300000;
- const int kNewMaxBitrateBps = 1234567;
-
ChangingNetworkRouteTest()
- : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
- call_(nullptr) {}
+ : EndToEndTest(test::CallTest::kDefaultTimeoutMs), call_(nullptr) {
+ EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionTransportSequenceNumber, kExtensionId));
+ }
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
call_ = sender_call;
}
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+ (*receive_configs)[0].rtp.transport_cc = true;
+ }
+
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override {
+ send_config->rtp.extensions.clear();
+ send_config->rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+ (*receive_configs)[0].rtp.extensions.clear();
+ (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+ (*receive_configs)[0].rtp.transport_cc = true;
+ }
stefan-webrtc 2016/09/27 10:58:55 Changes to enable send-side BWE for this test. Thi
+
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (call_->GetStats().send_bandwidth_bps > kStartBitrateBps) {
observation_complete_.Set();

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