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Fix potential deadlock in VideoSendStreamTests.

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The synchronously waiting SendTask() helper method should never be
called from within OnSendRtp() as that risks a deadlock with the
shutdown of the test.

This CL reverts an earlier disabling which did not correctly identify
the root cause.

Bug: webrtc:13291
Change-Id: Ia3c3417e0cbfb7011bb2439a52f976b946adad78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235721
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35244}
This commit is contained in:
Erik Språng 2021-10-20 10:28:59 +02:00 committed by WebRTC LUCI CQ
parent 42b983a606
commit 5316a061ca
1 changed files with 68 additions and 72 deletions
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140
video/video_send_stream_tests.cc
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@ -1458,14 +1458,7 @@ TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
//
// Note that the test starts at "high" bitrate and does not ramp up to "higher"
// bitrate since no receiver block or remb is sent in the initial phase.
// TODO(bugs.webrtc.org/13294): Fix flakiness or replace with scenario test.
#if defined(WEBRTC_MAC)
#define MAYBE_MinTransmitBitrateRespectsRemb \
DISABLED_MinTransmitBitrateRespectsRemb
#else
#define MAYBE_MinTransmitBitrateRespectsRemb MinTransmitBitrateRespectsRemb
#endif
TEST_F(VideoSendStreamTest, MAYBE_MinTransmitBitrateRespectsRemb) {
TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
static const int kMinTransmitBitrateBps = 400000;
static const int kHighBitrateBps = 150000;
static const int kRembBitrateBps = 80000;
@ -1490,27 +1483,29 @@ TEST_F(VideoSendStreamTest, MAYBE_MinTransmitBitrateRespectsRemb) {
return DROP_PACKET;
RtpPacket rtp_packet;
if (!rtp_packet.Parse(packet, length))
return DROP_PACKET;
RTC_CHECK(rtp_packet.Parse(packet, length));
const uint32_t ssrc = rtp_packet.Ssrc();
RTC_DCHECK(stream_);
VideoSendStream::Stats stats;
SendTask(RTC_FROM_HERE, task_queue_,
[&]() { stats = stream_->GetStats(); });
if (!stats.substreams.empty()) {
EXPECT_EQ(1u, stats.substreams.size());
int total_bitrate_bps =
stats.substreams.begin()->second.total_bitrate_bps;
test::PrintResult("bitrate_stats_", "min_transmit_bitrate_low_remb",
"bitrate_bps", static_cast<size_t>(total_bitrate_bps),
"bps", false);
if (total_bitrate_bps > kHighBitrateBps) {
rtp_rtcp_->SetRemb(kRembBitrateBps, {rtp_packet.Ssrc()});
bitrate_capped_ = true;
} else if (bitrate_capped_ &&
total_bitrate_bps < kRembRespectedBitrateBps) {
observation_complete_.Set();
task_queue_->PostTask(ToQueuedTask([this, ssrc]() {
VideoSendStream::Stats stats = stream_->GetStats();
if (!stats.substreams.empty()) {
EXPECT_EQ(1u, stats.substreams.size());
int total_bitrate_bps =
stats.substreams.begin()->second.total_bitrate_bps;
test::PrintResult(
"bitrate_stats_", "min_transmit_bitrate_low_remb", "bitrate_bps",
static_cast<size_t>(total_bitrate_bps), "bps", false);
if (total_bitrate_bps > kHighBitrateBps) {
rtp_rtcp_->SetRemb(kRembBitrateBps, {ssrc});
bitrate_capped_ = true;
} else if (bitrate_capped_ &&
total_bitrate_bps < kRembRespectedBitrateBps) {
observation_complete_.Set();
}
}
}
}));
// Packets don't have to be delivered since the test is the receiver.
return DROP_PACKET;
}
@ -3893,54 +3888,55 @@ class ContentSwitchTest : public test::SendTest {
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
task_queue_->PostTask(ToQueuedTask([this]() {
MutexLock lock(&mutex_);
auto internal_send_peer = test::VideoSendStreamPeer(send_stream_);
float pacing_factor =
internal_send_peer.GetPacingFactorOverride().value_or(0.0f);
float expected_pacing_factor = 1.1; // Strict pacing factor.
VideoSendStream::Stats stats;
SendTask(RTC_FROM_HERE, task_queue_,
[&stats, stream = send_stream_]() { stats = stream->GetStats(); });
if (stats.content_type == webrtc::VideoContentType::SCREENSHARE) {
expected_pacing_factor = 1.0f; // Currently used pacing factor in ALR.
}
EXPECT_NEAR(expected_pacing_factor, pacing_factor, 1e-6);
// Wait until at least kMinPacketsToSend packets to be sent, so that
// some frames would be encoded.
if (++packets_sent_ < kMinPacketsToSend)
return SEND_PACKET;
if (state_ != StreamState::kAfterSwitchBack) {
// We've sent kMinPacketsToSend packets, switch the content type and move
// move to the next state.
// Note that we need to recreate the stream if changing content type.
packets_sent_ = 0;
if (encoder_config_.content_type ==
VideoEncoderConfig::ContentType::kRealtimeVideo) {
encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
} else {
encoder_config_.content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
auto internal_send_peer = test::VideoSendStreamPeer(send_stream_);
float pacing_factor =
internal_send_peer.GetPacingFactorOverride().value_or(0.0f);
float expected_pacing_factor = 1.1; // Strict pacing factor.
VideoSendStream::Stats stats = send_stream_->GetStats();
if (stats.content_type == webrtc::VideoContentType::SCREENSHARE) {
expected_pacing_factor = 1.0f; // Currently used pacing factor in ALR.
}
switch (state_) {
case StreamState::kBeforeSwitch:
state_ = StreamState::kInScreenshare;
break;
case StreamState::kInScreenshare:
state_ = StreamState::kAfterSwitchBack;
break;
case StreamState::kAfterSwitchBack:
RTC_NOTREACHED();
break;
}
content_switch_event_.Set();
return SEND_PACKET;
}
observation_complete_.Set();
EXPECT_NEAR(expected_pacing_factor, pacing_factor, 1e-6);
// Wait until at least kMinPacketsToSend packets to be sent, so that
// some frames would be encoded.
if (++packets_sent_ < kMinPacketsToSend)
return;
if (state_ != StreamState::kAfterSwitchBack) {
// We've sent kMinPacketsToSend packets, switch the content type and
// move move to the next state. Note that we need to recreate the stream
// if changing content type.
packets_sent_ = 0;
if (encoder_config_.content_type ==
VideoEncoderConfig::ContentType::kRealtimeVideo) {
encoder_config_.content_type =
VideoEncoderConfig::ContentType::kScreen;
} else {
encoder_config_.content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
}
switch (state_) {
case StreamState::kBeforeSwitch:
state_ = StreamState::kInScreenshare;
break;
case StreamState::kInScreenshare:
state_ = StreamState::kAfterSwitchBack;
break;
case StreamState::kAfterSwitchBack:
RTC_NOTREACHED();
break;
}
content_switch_event_.Set();
return;
}
observation_complete_.Set();
}));
return SEND_PACKET;
}