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Use a new instance of RTP stack for each test.

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- Reusing RTP stack may have contributed to some flakiness as
  the previous state could have persisted to new test being performed.

Bug: webrtc:13241
Change-Id: Idf70b56bd3377bc99321fddf7191d7a72c37b085
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237540
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35336}
This commit is contained in:
Tim Na 2021-11-09 11:54:29 -08:00 committed by WebRTC LUCI CQ
parent c86e1c2e70
commit 1d73243466
1 changed files with 2 additions and 1 deletions
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3
audio/voip/test/audio_egress_unittest.cc
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@ -57,7 +57,6 @@ class AudioEgressTest : public ::testing::Test {
AudioEgressTest()
: fake_clock_(kStartTime), wave_generator_(1000.0, kAudioLevel) {
rtp_rtcp_ = CreateRtpStack(&fake_clock_, &transport_, kRemoteSsrc);
task_queue_factory_ = CreateDefaultTaskQueueFactory();
encoder_factory_ = CreateBuiltinAudioEncoderFactory();
}
@ -65,6 +64,7 @@ class AudioEgressTest : public ::testing::Test {
// Prepare test on audio egress by using PCMu codec with specific
// sequence number and its status to be running.
void SetUp() override {
rtp_rtcp_ = CreateRtpStack(&fake_clock_, &transport_, kRemoteSsrc);
egress_ = std::make_unique<AudioEgress>(rtp_rtcp_.get(), &fake_clock_,
task_queue_factory_.get());
constexpr int kPcmuPayload = 0;
@ -81,6 +81,7 @@ class AudioEgressTest : public ::testing::Test {
egress_->StopSend();
rtp_rtcp_->SetSendingStatus(false);
egress_.reset();
rtp_rtcp_.reset();
}
// Create an audio frame prepared for pcmu encoding. Timestamp is